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/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 2013, Digium, Inc.
*
* Joshua Colp <jcolp@digium.com>
*
* See http://www.asterisk.org for more information about
* the Asterisk project. Please do not directly contact
* any of the maintainers of this project for assistance;
* the project provides a web site, mailing lists and IRC
* channels for your use.
*
* This program is free software, distributed under the terms of
* the GNU General Public License Version 2. See the LICENSE file
* at the top of the source tree.
*/
/*** MODULEINFO
<depend>pjproject</depend>
<depend>res_pjsip</depend>
<use type="module">res_statsd</use>
<support_level>core</support_level>
***/
#include "asterisk.h"
#include <pjsip.h>
#include <pjsip_ua.h>
#include "asterisk/res_pjsip.h"
#include "asterisk/res_pjsip_cli.h"
#include "asterisk/module.h"
#include "asterisk/taskprocessor.h"
#include "asterisk/cli.h"
#include "asterisk/stasis_system.h"
#include "asterisk/threadstorage.h"
#include "asterisk/threadpool.h"
#include "asterisk/statsd.h"
res_pjsip: AMI commands and events. 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/ ........ Merged revisions 403131 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403133 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-11-23 17:26:57 +00:00
#include "res_pjsip/include/res_pjsip_private.h"
res_pjsip: Implement additional SIP RFCs for Google Voice trunk compatability This change implements a few different generic things which were brought on by Google Voice SIP. 1. The concept of flow transports have been introduced. These are configurable transports in pjsip.conf which can be used to reference a flow of signaling to a target. These have runtime configuration that can be changed by the signaling itself (such as Service-Routes and P-Preferred-Identity). When used these guarantee an individual connection (in the case of TCP or TLS) even if multiple flow transports exist to the same target. 2. Service-Routes (RFC 3608) support has been added to the outbound registration module which when received will be stored on the flow transport and used for requests referencing it. 3. P-Associated-URI / P-Preferred-Identity (RFC 3325) support has been added to the outbound registration module. If a P-Associated-URI header is received it will be used on requests as the P-Preferred-Identity. 4. Configurable outbound extension support has been added to the outbound registration module. When set the extension will be placed in the Supported header. 5. Header parameters can now be configured on an outbound registration which will be placed in the Contact header. 6. Google specific OAuth / Bearer token authentication (draft-ietf-sipcore-sip-authn-02) has been added to the outbound registration module. All functionality changes are controlled by pjsip.conf configuration options and do not affect non-configured pjsip endpoints otherwise. ASTERISK-27971 #close Change-Id: Id214c2d1c550a41fcf564b7df8f3da7be565bd58
2018-07-18 12:45:26 +00:00
#include "asterisk/vector.h"
#include "asterisk/pbx.h"
/*** DOCUMENTATION
<configInfo name="res_pjsip_outbound_registration" language="en_US">
<synopsis>SIP resource for outbound registrations</synopsis>
<description><para>
<emphasis>Outbound Registration</emphasis>
</para>
<para>This module allows <literal>res_pjsip</literal> to register to other SIP servers.</para>
</description>
<configFile name="pjsip.conf">
<configObject name="registration">
<synopsis>The configuration for outbound registration</synopsis>
<description><para>
Registration is <emphasis>COMPLETELY</emphasis> separate from the rest of
<literal>pjsip.conf</literal>. A minimal configuration consists of
setting a <literal>server_uri</literal> and a <literal>client_uri</literal>.
</para></description>
<configOption name="auth_rejection_permanent" default="yes">
<synopsis>Determines whether failed authentication challenges are treated
as permanent failures.</synopsis>
<description><para>If this option is enabled and an authentication challenge fails,
registration will not be attempted again until the configuration is reloaded.</para></description>
</configOption>
<configOption name="client_uri">
<synopsis>Client SIP URI used when attemping outbound registration</synopsis>
<description><para>
This is the address-of-record for the outbound registration (i.e. the URI in
the To header of the REGISTER).</para>
<para>For registration with an ITSP, the client SIP URI may need to consist of
an account name or number and the provider's hostname for their registrar, e.g.
client_uri=1234567890@example.com. This may differ between providers.</para>
<para>For registration to generic registrars, the client SIP URI will depend
on networking specifics and configuration of the registrar.
</para></description>
</configOption>
<configOption name="contact_user" default="s">
<synopsis>Contact User to use in request. If this value is not set, this defaults to 's'</synopsis>
</configOption>
res_pjsip: Implement additional SIP RFCs for Google Voice trunk compatability This change implements a few different generic things which were brought on by Google Voice SIP. 1. The concept of flow transports have been introduced. These are configurable transports in pjsip.conf which can be used to reference a flow of signaling to a target. These have runtime configuration that can be changed by the signaling itself (such as Service-Routes and P-Preferred-Identity). When used these guarantee an individual connection (in the case of TCP or TLS) even if multiple flow transports exist to the same target. 2. Service-Routes (RFC 3608) support has been added to the outbound registration module which when received will be stored on the flow transport and used for requests referencing it. 3. P-Associated-URI / P-Preferred-Identity (RFC 3325) support has been added to the outbound registration module. If a P-Associated-URI header is received it will be used on requests as the P-Preferred-Identity. 4. Configurable outbound extension support has been added to the outbound registration module. When set the extension will be placed in the Supported header. 5. Header parameters can now be configured on an outbound registration which will be placed in the Contact header. 6. Google specific OAuth / Bearer token authentication (draft-ietf-sipcore-sip-authn-02) has been added to the outbound registration module. All functionality changes are controlled by pjsip.conf configuration options and do not affect non-configured pjsip endpoints otherwise. ASTERISK-27971 #close Change-Id: Id214c2d1c550a41fcf564b7df8f3da7be565bd58
2018-07-18 12:45:26 +00:00
<configOption name="contact_header_params">
<synopsis>Header parameters to place in the Contact header</synopsis>
</configOption>
<configOption name="expiration" default="3600">
<synopsis>Expiration time for registrations in seconds</synopsis>
</configOption>
<configOption name="max_retries" default="10">
<synopsis>Maximum number of registration attempts.</synopsis>
<description><para>
This sets the maximum number of registration attempts that are made before
stopping any further attempts. If set to 0 then upon failure no further attempts
are made.
</para></description>
</configOption>
<configOption name="security_negotiation" default="no">
<synopsis>The kind of security agreement negotiation to use. Currently, only mediasec is supported.</synopsis>
<description>
<enumlist>
<enum name="no" />
<enum name="mediasec" />
</enumlist>
</description>
</configOption>
<configOption name="security_mechanisms">
<synopsis>List of security mechanisms supported.</synopsis>
<description><para>
This is a comma-delimited list of security mechanisms to use. Each security mechanism
must be in the form defined by RFC 3329 section 2.2.
</para></description>
</configOption>
<configOption name="outbound_auth" default="">
<synopsis>Authentication object(s) to be used for outbound registrations.</synopsis>
<description><para>
This is a comma-delimited list of <replaceable>auth</replaceable>
sections defined in <filename>pjsip.conf</filename> used to respond
to outbound authentication challenges.</para>
<note><para>
Using the same auth section for inbound and outbound
authentication is not recommended. There is a difference in
meaning for an empty realm setting between inbound and outbound
authentication uses. See the auth realm description for details.
</para></note>
</description>
</configOption>
<configOption name="outbound_proxy" default="">
<synopsis>Full SIP URI of the outbound proxy used to send registrations</synopsis>
</configOption>
<configOption name="max_random_initial_delay" default="10">
<synopsis>Maximum interval in seconds for which an initial registration may be randomly delayed</synopsis>
<description>
<para>By default, registrations are randomly delayed by a small amount to prevent
too many registrations from being made simultaneously.</para>
<para>Depending on your system usage, it may be desirable to set this to a smaller
or larger value to have fine grained control over the size of this random delay.</para>
</description>
</configOption>
<configOption name="retry_interval" default="60">
<synopsis>Interval in seconds between retries if outbound registration is unsuccessful</synopsis>
</configOption>
<configOption name="forbidden_retry_interval" default="0">
<synopsis>Interval used when receiving a 403 Forbidden response.</synopsis>
<description><para>
If a 403 Forbidden is received, chan_pjsip will wait
<replaceable>forbidden_retry_interval</replaceable> seconds before
attempting registration again. If 0 is specified, chan_pjsip will not
retry after receiving a 403 Forbidden response. Setting this to a non-zero
value goes against a "SHOULD NOT" in RFC3261, but can be used to work around
buggy registrars.
</para></description>
</configOption>
<configOption name="fatal_retry_interval" default="0">
<synopsis>Interval used when receiving a Fatal response.</synopsis>
<description><para>
If a fatal response is received, chan_pjsip will wait
<replaceable>fatal_retry_interval</replaceable> seconds before
attempting registration again. If 0 is specified, chan_pjsip will not
retry after receiving a fatal (non-temporary 4xx, 5xx, 6xx) response.
Setting this to a non-zero value may go against a "SHOULD NOT" in RFC3261,
but can be used to work around buggy registrars.</para>
<note><para>if also set the <replaceable>forbidden_retry_interval</replaceable>
takes precedence over this one when a 403 is received.
Also, if <replaceable>auth_rejection_permanent</replaceable> equals 'yes' then
a 401 and 407 become subject to this retry interval.</para></note>
</description>
</configOption>
<configOption name="server_uri">
<synopsis>SIP URI of the server to register against</synopsis>
<description><para>
This is the URI at which to find the registrar to send the outbound REGISTER. This URI
is used as the request URI of the outbound REGISTER request from Asterisk.</para>
<para>For registration with an ITSP, the setting may often be just the domain of
the registrar, e.g. sip:sip.example.com.
</para></description>
</configOption>
<configOption name="transport">
<synopsis>Transport used for outbound authentication</synopsis>
<description>
<note><para>A <replaceable>transport</replaceable> configured in
<literal>pjsip.conf</literal>. As with other <literal>res_pjsip</literal> modules, this will use the first available transport of the appropriate type if unconfigured.</para></note>
</description>
</configOption>
<configOption name="line">
<synopsis>Whether to add a 'line' parameter to the Contact for inbound call matching</synopsis>
<description><para>
When enabled this option will cause a 'line' parameter to be added to the Contact
header placed into the outgoing registration request. If the remote server sends a call
this line parameter will be used to establish a relationship to the outbound registration,
ultimately causing the configured endpoint to be used.
</para></description>
</configOption>
<configOption name="endpoint">
<synopsis>Endpoint to use for incoming related calls</synopsis>
<description><para>
When line support is enabled this configured endpoint name is used for incoming calls
that are related to the outbound registration.
</para></description>
</configOption>
<configOption name="type">
<synopsis>Must be of type 'registration'.</synopsis>
</configOption>
<configOption name="support_path">
res_pjsip: Implement additional SIP RFCs for Google Voice trunk compatability This change implements a few different generic things which were brought on by Google Voice SIP. 1. The concept of flow transports have been introduced. These are configurable transports in pjsip.conf which can be used to reference a flow of signaling to a target. These have runtime configuration that can be changed by the signaling itself (such as Service-Routes and P-Preferred-Identity). When used these guarantee an individual connection (in the case of TCP or TLS) even if multiple flow transports exist to the same target. 2. Service-Routes (RFC 3608) support has been added to the outbound registration module which when received will be stored on the flow transport and used for requests referencing it. 3. P-Associated-URI / P-Preferred-Identity (RFC 3325) support has been added to the outbound registration module. If a P-Associated-URI header is received it will be used on requests as the P-Preferred-Identity. 4. Configurable outbound extension support has been added to the outbound registration module. When set the extension will be placed in the Supported header. 5. Header parameters can now be configured on an outbound registration which will be placed in the Contact header. 6. Google specific OAuth / Bearer token authentication (draft-ietf-sipcore-sip-authn-02) has been added to the outbound registration module. All functionality changes are controlled by pjsip.conf configuration options and do not affect non-configured pjsip endpoints otherwise. ASTERISK-27971 #close Change-Id: Id214c2d1c550a41fcf564b7df8f3da7be565bd58
2018-07-18 12:45:26 +00:00
<synopsis>Enables advertising SIP Path support for outbound REGISTER requests.</synopsis>
<description><para>
When this option is enabled, outbound REGISTER requests will advertise
support for Path headers so that intervening proxies can add to the Path
header as necessary.
</para></description>
</configOption>
res_pjsip: Implement additional SIP RFCs for Google Voice trunk compatability This change implements a few different generic things which were brought on by Google Voice SIP. 1. The concept of flow transports have been introduced. These are configurable transports in pjsip.conf which can be used to reference a flow of signaling to a target. These have runtime configuration that can be changed by the signaling itself (such as Service-Routes and P-Preferred-Identity). When used these guarantee an individual connection (in the case of TCP or TLS) even if multiple flow transports exist to the same target. 2. Service-Routes (RFC 3608) support has been added to the outbound registration module which when received will be stored on the flow transport and used for requests referencing it. 3. P-Associated-URI / P-Preferred-Identity (RFC 3325) support has been added to the outbound registration module. If a P-Associated-URI header is received it will be used on requests as the P-Preferred-Identity. 4. Configurable outbound extension support has been added to the outbound registration module. When set the extension will be placed in the Supported header. 5. Header parameters can now be configured on an outbound registration which will be placed in the Contact header. 6. Google specific OAuth / Bearer token authentication (draft-ietf-sipcore-sip-authn-02) has been added to the outbound registration module. All functionality changes are controlled by pjsip.conf configuration options and do not affect non-configured pjsip endpoints otherwise. ASTERISK-27971 #close Change-Id: Id214c2d1c550a41fcf564b7df8f3da7be565bd58
2018-07-18 12:45:26 +00:00
<configOption name="support_outbound">
<synopsis>Enables advertising SIP Outbound support (RFC5626) for outbound REGISTER requests.</synopsis>
</configOption>
<configOption name="user_agent">
<synopsis>Overrides the User-Agent header that should be used for outbound REGISTER requests.</synopsis>
</configOption>
</configObject>
</configFile>
</configInfo>
<manager name="PJSIPUnregister" language="en_US">
<synopsis>
Unregister an outbound registration.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Registration" required="true">
<para>The outbound registration to unregister or '*all' to unregister them all.</para>
</parameter>
</syntax>
<description>
<para>
Unregisters the specified (or all) outbound registration(s) and stops future registration attempts.
Call PJSIPRegister to start registration and schedule re-registrations according to configuration.
</para>
</description>
</manager>
<manager name="PJSIPRegister" language="en_US">
<synopsis>
Register an outbound registration.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Registration" required="true">
<para>The outbound registration to register or '*all' to register them all.</para>
</parameter>
</syntax>
<description>
<para>
Unregisters the specified (or all) outbound registration(s) then starts registration and schedules re-registrations
according to configuration.
</para>
</description>
</manager>
res_pjsip: AMI commands and events. 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/ ........ Merged revisions 403131 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403133 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-11-23 17:26:57 +00:00
<manager name="PJSIPShowRegistrationsOutbound" language="en_US">
<synopsis>
Lists PJSIP outbound registrations.
</synopsis>
<syntax />
<description>
<para>
In response <literal>OutboundRegistrationDetail</literal> events showing configuration and status
information are raised for each outbound registration object. <literal>AuthDetail</literal>
events are raised for each associated auth object as well. Once all events are completed an
<literal>OutboundRegistrationDetailComplete</literal> is issued.
</para>
</description>
</manager>
***/
res_pjsip: Implement additional SIP RFCs for Google Voice trunk compatability This change implements a few different generic things which were brought on by Google Voice SIP. 1. The concept of flow transports have been introduced. These are configurable transports in pjsip.conf which can be used to reference a flow of signaling to a target. These have runtime configuration that can be changed by the signaling itself (such as Service-Routes and P-Preferred-Identity). When used these guarantee an individual connection (in the case of TCP or TLS) even if multiple flow transports exist to the same target. 2. Service-Routes (RFC 3608) support has been added to the outbound registration module which when received will be stored on the flow transport and used for requests referencing it. 3. P-Associated-URI / P-Preferred-Identity (RFC 3325) support has been added to the outbound registration module. If a P-Associated-URI header is received it will be used on requests as the P-Preferred-Identity. 4. Configurable outbound extension support has been added to the outbound registration module. When set the extension will be placed in the Supported header. 5. Header parameters can now be configured on an outbound registration which will be placed in the Contact header. 6. Google specific OAuth / Bearer token authentication (draft-ietf-sipcore-sip-authn-02) has been added to the outbound registration module. All functionality changes are controlled by pjsip.conf configuration options and do not affect non-configured pjsip endpoints otherwise. ASTERISK-27971 #close Change-Id: Id214c2d1c550a41fcf564b7df8f3da7be565bd58
2018-07-18 12:45:26 +00:00
/* forward declarations */
static int set_outbound_initial_authentication_credentials(pjsip_regc *regc,
const struct ast_sip_auth_vector *auth_vector);
/*! \brief Some thread local storage used to determine if the running thread invoked the callback */
AST_THREADSTORAGE(register_callback_invoked);
/*! \brief Amount of buffer time (in seconds) before expiration that we re-register at */
#define REREGISTER_BUFFER_TIME 10
/*! \brief Size of the buffer for creating a unique string for the line */
#define LINE_PARAMETER_SIZE 8
/*! \brief Various states that an outbound registration may be in */
enum sip_outbound_registration_status {
/*! \brief Currently unregistered */
SIP_REGISTRATION_UNREGISTERED = 0,
/*! \brief Registered, yay! */
SIP_REGISTRATION_REGISTERED,
/*! \brief Registration was rejected, but response was temporal */
SIP_REGISTRATION_REJECTED_TEMPORARY,
/*! \brief Registration was rejected, permanently */
SIP_REGISTRATION_REJECTED_PERMANENT,
res_pjsip_outbound_registration.c: Fix handle_client_state_destruction() refs * handle_client_state_destruction() must always be passed a ref to client_state because it will always unref client_state. handle_registration_response() was not passing a client_state ref. * Made the final un-REGISTER message get sent normally using the pjproject register control structure in handle_client_state_destruction(). The previous code attempted to short circuit the response handling for the module to unload. That doesn't work for a couple reasons. One, pjsip_regc_send() may call the registered callback before it returns and unbalance the client_state ref count. Two, the registered callback handles any authentication for the un-REGISTER message. * Made the distinction between internal registration state and external registration status with sip_outbound_registration_status_str(). This is necessary to avoid altering documented AMI messages with internal changes. * Removed references to client_state->client outside of the serializer thread. When handle_client_state_destruction() destroys the pjproject register control structure that memory is freed and cannot be referenced anymore. These accesses were to provide information for debug and off-nominal warning messages. * In sip_outbound_registration_timer_cb() you should not access entry->id after unrefing client_state because the passed in entry is normally pointing to the timer entry in the client_state object. ASTERISK-24907 Reported by: Kevin Harwell Change-Id: Ia7b446d8644b6b4550ef5bea49527671de65183f
2015-06-16 20:06:22 +00:00
/*! \brief Registration is stopping. */
SIP_REGISTRATION_STOPPING,
/*! \brief Registration has been stopped */
SIP_REGISTRATION_STOPPED,
};
res_pjsip_outbound_registration.c: Fix handle_client_state_destruction() refs * handle_client_state_destruction() must always be passed a ref to client_state because it will always unref client_state. handle_registration_response() was not passing a client_state ref. * Made the final un-REGISTER message get sent normally using the pjproject register control structure in handle_client_state_destruction(). The previous code attempted to short circuit the response handling for the module to unload. That doesn't work for a couple reasons. One, pjsip_regc_send() may call the registered callback before it returns and unbalance the client_state ref count. Two, the registered callback handles any authentication for the un-REGISTER message. * Made the distinction between internal registration state and external registration status with sip_outbound_registration_status_str(). This is necessary to avoid altering documented AMI messages with internal changes. * Removed references to client_state->client outside of the serializer thread. When handle_client_state_destruction() destroys the pjproject register control structure that memory is freed and cannot be referenced anymore. These accesses were to provide information for debug and off-nominal warning messages. * In sip_outbound_registration_timer_cb() you should not access entry->id after unrefing client_state because the passed in entry is normally pointing to the timer entry in the client_state object. ASTERISK-24907 Reported by: Kevin Harwell Change-Id: Ia7b446d8644b6b4550ef5bea49527671de65183f
2015-06-16 20:06:22 +00:00
/*!
* \internal
* \brief Convert the internal registration state to an external status string.
* \since 13.5.0
*
* \param state Current outbound registration state.
*
* \return External registration status string.
*/
static const char *sip_outbound_registration_status_str(enum sip_outbound_registration_status state)
{
const char *str;
str = "Unregistered";
switch (state) {
case SIP_REGISTRATION_STOPPING:
case SIP_REGISTRATION_STOPPED:
case SIP_REGISTRATION_UNREGISTERED:
break;
case SIP_REGISTRATION_REGISTERED:
str = "Registered";
break;
case SIP_REGISTRATION_REJECTED_TEMPORARY:
case SIP_REGISTRATION_REJECTED_PERMANENT:
str = "Rejected";
break;
}
return str;
}
/*! \brief Outbound registration information */
struct sip_outbound_registration {
/*! \brief Sorcery object details */
SORCERY_OBJECT(details);
/*! \brief Stringfields */
AST_DECLARE_STRING_FIELDS(
/*! \brief URI for the registrar */
AST_STRING_FIELD(server_uri);
/*! \brief URI for the AOR */
AST_STRING_FIELD(client_uri);
/*! \brief Optional user for contact header */
AST_STRING_FIELD(contact_user);
/*! \brief Optional header parameters for contact */
res_pjsip: Implement additional SIP RFCs for Google Voice trunk compatability This change implements a few different generic things which were brought on by Google Voice SIP. 1. The concept of flow transports have been introduced. These are configurable transports in pjsip.conf which can be used to reference a flow of signaling to a target. These have runtime configuration that can be changed by the signaling itself (such as Service-Routes and P-Preferred-Identity). When used these guarantee an individual connection (in the case of TCP or TLS) even if multiple flow transports exist to the same target. 2. Service-Routes (RFC 3608) support has been added to the outbound registration module which when received will be stored on the flow transport and used for requests referencing it. 3. P-Associated-URI / P-Preferred-Identity (RFC 3325) support has been added to the outbound registration module. If a P-Associated-URI header is received it will be used on requests as the P-Preferred-Identity. 4. Configurable outbound extension support has been added to the outbound registration module. When set the extension will be placed in the Supported header. 5. Header parameters can now be configured on an outbound registration which will be placed in the Contact header. 6. Google specific OAuth / Bearer token authentication (draft-ietf-sipcore-sip-authn-02) has been added to the outbound registration module. All functionality changes are controlled by pjsip.conf configuration options and do not affect non-configured pjsip endpoints otherwise. ASTERISK-27971 #close Change-Id: Id214c2d1c550a41fcf564b7df8f3da7be565bd58
2018-07-18 12:45:26 +00:00
AST_STRING_FIELD(contact_header_params);
/*! \brief Explicit transport to use for registration */
AST_STRING_FIELD(transport);
/*! \brief Outbound proxy to use */
AST_STRING_FIELD(outbound_proxy);
/*! \brief Endpoint to use for related incoming calls */
AST_STRING_FIELD(endpoint);
/*! \brief User-Agent to use when sending the REGISTER */
AST_STRING_FIELD(user_agent);
);
/*! \brief Requested expiration time */
unsigned int expiration;
/*! \brief Maximum random initial delay interval for initial registrations */
unsigned int max_random_initial_delay;
/*! \brief Interval at which retries should occur for temporal responses */
unsigned int retry_interval;
/*! \brief Interval at which retries should occur for permanent responses */
unsigned int forbidden_retry_interval;
/*! \brief Interval at which retries should occur for all permanent responses */
unsigned int fatal_retry_interval;
/*! \brief Treat authentication challenges that we cannot handle as permanent failures */
unsigned int auth_rejection_permanent;
/*! \brief Maximum number of retries permitted */
unsigned int max_retries;
/*! \brief Whether to add a line parameter to the outbound Contact or not */
unsigned int line;
/*! \brief Type of security negotiation to use (RFC 3329). */
enum ast_sip_security_negotiation security_negotiation;
/*! \brief Client security mechanisms (RFC 3329). */
struct ast_sip_security_mechanism_vector security_mechanisms;
/*! \brief Configured authentication credentials */
struct ast_sip_auth_vector outbound_auths;
/*! \brief Whether Path support is enabled */
unsigned int support_path;
res_pjsip: Implement additional SIP RFCs for Google Voice trunk compatability This change implements a few different generic things which were brought on by Google Voice SIP. 1. The concept of flow transports have been introduced. These are configurable transports in pjsip.conf which can be used to reference a flow of signaling to a target. These have runtime configuration that can be changed by the signaling itself (such as Service-Routes and P-Preferred-Identity). When used these guarantee an individual connection (in the case of TCP or TLS) even if multiple flow transports exist to the same target. 2. Service-Routes (RFC 3608) support has been added to the outbound registration module which when received will be stored on the flow transport and used for requests referencing it. 3. P-Associated-URI / P-Preferred-Identity (RFC 3325) support has been added to the outbound registration module. If a P-Associated-URI header is received it will be used on requests as the P-Preferred-Identity. 4. Configurable outbound extension support has been added to the outbound registration module. When set the extension will be placed in the Supported header. 5. Header parameters can now be configured on an outbound registration which will be placed in the Contact header. 6. Google specific OAuth / Bearer token authentication (draft-ietf-sipcore-sip-authn-02) has been added to the outbound registration module. All functionality changes are controlled by pjsip.conf configuration options and do not affect non-configured pjsip endpoints otherwise. ASTERISK-27971 #close Change-Id: Id214c2d1c550a41fcf564b7df8f3da7be565bd58
2018-07-18 12:45:26 +00:00
/*! \brief Whether Outbound support is enabled */
unsigned int support_outbound;
};
/*! \brief Outbound registration client state information (persists for lifetime of regc) */
struct sip_outbound_registration_client_state {
res_pjsip_outbound_registration.c: Fix handle_client_state_destruction() refs * handle_client_state_destruction() must always be passed a ref to client_state because it will always unref client_state. handle_registration_response() was not passing a client_state ref. * Made the final un-REGISTER message get sent normally using the pjproject register control structure in handle_client_state_destruction(). The previous code attempted to short circuit the response handling for the module to unload. That doesn't work for a couple reasons. One, pjsip_regc_send() may call the registered callback before it returns and unbalance the client_state ref count. Two, the registered callback handles any authentication for the un-REGISTER message. * Made the distinction between internal registration state and external registration status with sip_outbound_registration_status_str(). This is necessary to avoid altering documented AMI messages with internal changes. * Removed references to client_state->client outside of the serializer thread. When handle_client_state_destruction() destroys the pjproject register control structure that memory is freed and cannot be referenced anymore. These accesses were to provide information for debug and off-nominal warning messages. * In sip_outbound_registration_timer_cb() you should not access entry->id after unrefing client_state because the passed in entry is normally pointing to the timer entry in the client_state object. ASTERISK-24907 Reported by: Kevin Harwell Change-Id: Ia7b446d8644b6b4550ef5bea49527671de65183f
2015-06-16 20:06:22 +00:00
/*! \brief Current state of this registration */
enum sip_outbound_registration_status status;
res_pjsip_outbound_registration.c: Fix handle_client_state_destruction() refs * handle_client_state_destruction() must always be passed a ref to client_state because it will always unref client_state. handle_registration_response() was not passing a client_state ref. * Made the final un-REGISTER message get sent normally using the pjproject register control structure in handle_client_state_destruction(). The previous code attempted to short circuit the response handling for the module to unload. That doesn't work for a couple reasons. One, pjsip_regc_send() may call the registered callback before it returns and unbalance the client_state ref count. Two, the registered callback handles any authentication for the un-REGISTER message. * Made the distinction between internal registration state and external registration status with sip_outbound_registration_status_str(). This is necessary to avoid altering documented AMI messages with internal changes. * Removed references to client_state->client outside of the serializer thread. When handle_client_state_destruction() destroys the pjproject register control structure that memory is freed and cannot be referenced anymore. These accesses were to provide information for debug and off-nominal warning messages. * In sip_outbound_registration_timer_cb() you should not access entry->id after unrefing client_state because the passed in entry is normally pointing to the timer entry in the client_state object. ASTERISK-24907 Reported by: Kevin Harwell Change-Id: Ia7b446d8644b6b4550ef5bea49527671de65183f
2015-06-16 20:06:22 +00:00
/*!
* \brief Outbound registration client
* \note May only be accessed within the serializer thread
* because it might get destroyed and set to NULL for
* module unload.
*/
pjsip_regc *client;
res_pjsip_outbound_registration: Fix SRV failover on timeout In order to retry outbound registrations for some situations, we need access to the tdata from the original request. For instance, for 401/407 responses we need it to properly construct the subsequent request with the authentication. We also need it if we're iterating over a DNS SRV response record set so we can skip entries we've already tried. We've been getting the tdata from the server response rdata and transaction but that only works for the failures where there was actually a response (4XX, 5XX, etc). For timeouts there's no response and therefore no rdata or transaction from which to get the tdata. When processing a single A/AAAA record for a server, this wasn't an issue as we just retried that same server after the retry timer expired. If we got an SRV record set for the server though, without the state from the tdata, we just kept trying the first entry in the set repeatedly instead of skipping to the next one in the list. * Added a "last_tdata" member to the client state structure to keep track of the sent tdata. * Updated registration_client_send() to save the tdata it used into the client_state. * Updated sip_outbound_registration_response_cb() to use the tdata saved in client_state when we don't get a response from the server. We still use the tdata from the transaction when we DO get a response from the server so we can properly handle 4XX responses where our new request depends on it. General note on timeouts: Although res_pjsip_outbound_registration skips to the next record immediately when a timeout occurs during SRV set traversal, it's pjproject that determines how long to wait before a timeout is declared. As with other SIP message types, pjproject will continue trying the same server at an interval specified by "timer_t1" until "timer_b" expires. Both of those timers are set in the pjsip.conf "system" section. ASTERISK-28746 Change-Id: I199b8274392d17661dd3ce3b4d69a3968368fa06
2020-02-13 19:39:58 +00:00
/*!
* \brief Last tdata sent
* We need the original tdata to resend a request on auth failure
* or timeout. On an auth failure, we use the original tdata
* to initialize the new tdata for the authorized response. On a timeout
* we need it to skip failed SRV entries if any.
*/
pjsip_tx_data *last_tdata;
/*! \brief Timer entry for retrying on temporal responses */
pj_timer_entry timer;
/*! \brief Optional line parameter placed into Contact */
char line[LINE_PARAMETER_SIZE];
/*! \brief Current number of retries */
unsigned int retries;
/*! \brief Maximum number of retries permitted */
unsigned int max_retries;
/*! \brief Interval at which retries should occur for temporal responses */
unsigned int retry_interval;
/*! \brief Interval at which retries should occur for permanent responses */
unsigned int forbidden_retry_interval;
/*! \brief Interval at which retries should occur for all permanent responses */
unsigned int fatal_retry_interval;
/*! \brief Treat authentication challenges that we cannot handle as permanent failures */
unsigned int auth_rejection_permanent;
/*! \brief Determines whether SIP Path support should be advertised */
unsigned int support_path;
res_pjsip: Implement additional SIP RFCs for Google Voice trunk compatability This change implements a few different generic things which were brought on by Google Voice SIP. 1. The concept of flow transports have been introduced. These are configurable transports in pjsip.conf which can be used to reference a flow of signaling to a target. These have runtime configuration that can be changed by the signaling itself (such as Service-Routes and P-Preferred-Identity). When used these guarantee an individual connection (in the case of TCP or TLS) even if multiple flow transports exist to the same target. 2. Service-Routes (RFC 3608) support has been added to the outbound registration module which when received will be stored on the flow transport and used for requests referencing it. 3. P-Associated-URI / P-Preferred-Identity (RFC 3325) support has been added to the outbound registration module. If a P-Associated-URI header is received it will be used on requests as the P-Preferred-Identity. 4. Configurable outbound extension support has been added to the outbound registration module. When set the extension will be placed in the Supported header. 5. Header parameters can now be configured on an outbound registration which will be placed in the Contact header. 6. Google specific OAuth / Bearer token authentication (draft-ietf-sipcore-sip-authn-02) has been added to the outbound registration module. All functionality changes are controlled by pjsip.conf configuration options and do not affect non-configured pjsip endpoints otherwise. ASTERISK-27971 #close Change-Id: Id214c2d1c550a41fcf564b7df8f3da7be565bd58
2018-07-18 12:45:26 +00:00
/*! \brief Determines whether SIP Outbound support should be advertised */
unsigned int support_outbound;
/*! \brief Type of security negotiation to use (RFC 3329). */
enum ast_sip_security_negotiation security_negotiation;
/*! \brief Client security mechanisms (RFC 3329). */
struct ast_sip_security_mechanism_vector security_mechanisms;
/*! \brief Security mechanisms of the peer (RFC 3329). */
struct ast_sip_security_mechanism_vector server_security_mechanisms;
/*! CSeq number of last sent auth request. */
unsigned int auth_cseq;
/*! \brief Serializer for stuff and things */
struct ast_taskprocessor *serializer;
/*! \brief Configured authentication credentials */
struct ast_sip_auth_vector outbound_auths;
/*! \brief Registration should be destroyed after completion of transaction */
unsigned int destroy:1;
/*! \brief Non-zero if we have attempted sending a REGISTER with authentication */
unsigned int auth_attempted:1;
/*! \brief Status code of last response if we have tried to register before */
int last_status_code;
/*! \brief The name of the transport to be used for the registration */
char *transport_name;
/*! \brief The name of the registration sorcery object */
char *registration_name;
/*! \brief Expected time of registration lapse/expiration */
unsigned int registration_expires;
/*! \brief The value for the User-Agent header sent in requests */
char *user_agent;
};
/*! \brief Outbound registration state information (persists for lifetime that registration should exist) */
struct sip_outbound_registration_state {
/*! \brief Outbound registration configuration object */
struct sip_outbound_registration *registration;
/*! \brief Client state information */
struct sip_outbound_registration_client_state *client_state;
};
/*! Time needs to be long enough for a transaction to timeout if nothing replies. */
#define MAX_UNLOAD_TIMEOUT_TIME 35 /* Seconds */
/*! Shutdown group to monitor sip_outbound_registration_client_state serializers. */
static struct ast_serializer_shutdown_group *shutdown_group;
/*! \brief Default number of state container buckets */
#define DEFAULT_STATE_BUCKETS 53
static AO2_GLOBAL_OBJ_STATIC(current_states);
/*! subscription id for network change events */
static struct stasis_subscription *network_change_sub;
/*! \brief hashing function for state objects */
static int registration_state_hash(const void *obj, const int flags)
{
const struct sip_outbound_registration_state *object;
const char *key;
switch (flags & OBJ_SEARCH_MASK) {
case OBJ_SEARCH_KEY:
key = obj;
break;
case OBJ_SEARCH_OBJECT:
object = obj;
key = ast_sorcery_object_get_id(object->registration);
break;
default:
ast_assert(0);
return 0;
}
return ast_str_hash(key);
}
/*! \brief comparator function for state objects */
static int registration_state_cmp(void *obj, void *arg, int flags)
{
const struct sip_outbound_registration_state *object_left = obj;
const struct sip_outbound_registration_state *object_right = arg;
const char *right_key = arg;
int cmp;
switch (flags & OBJ_SEARCH_MASK) {
case OBJ_SEARCH_OBJECT:
right_key = ast_sorcery_object_get_id(object_right->registration);
/* Fall through */
case OBJ_SEARCH_KEY:
cmp = strcmp(ast_sorcery_object_get_id(object_left->registration), right_key);
break;
case OBJ_SEARCH_PARTIAL_KEY:
/* Not supported by container. */
ast_assert(0);
return 0;
default:
cmp = 0;
break;
}
if (cmp) {
return 0;
}
return CMP_MATCH;
}
static struct sip_outbound_registration_state *get_state(const char *id)
{
struct sip_outbound_registration_state *state = NULL;
struct ao2_container *states;
states = ao2_global_obj_ref(current_states);
if (states) {
state = ao2_find(states, id, OBJ_SEARCH_KEY);
ao2_ref(states, -1);
}
return state;
}
static struct ao2_container *get_registrations(void)
{
struct ao2_container *registrations = ast_sorcery_retrieve_by_fields(
ast_sip_get_sorcery(), "registration",
AST_RETRIEVE_FLAG_MULTIPLE | AST_RETRIEVE_FLAG_ALL, NULL);
return registrations;
}
/*! \brief Callback function for matching an outbound registration based on line */
static int line_identify_relationship(void *obj, void *arg, int flags)
{
struct sip_outbound_registration_state *state = obj;
pjsip_param *line = arg;
return !pj_strcmp2(&line->value, state->client_state->line) ? CMP_MATCH : 0;
}
static struct pjsip_param *get_uri_option_line(const void *uri)
{
static const pj_str_t LINE_STR = { "line", 4 };
return ast_sip_pjsip_uri_get_other_param((pjsip_uri *)uri, &LINE_STR);
}
/*! \brief Endpoint identifier which uses the 'line' parameter to establish a relationship to an outgoing registration */
static struct ast_sip_endpoint *line_identify(pjsip_rx_data *rdata)
{
pjsip_param *line;
RAII_VAR(struct ao2_container *, states, NULL, ao2_cleanup);
RAII_VAR(struct sip_outbound_registration_state *, state, NULL, ao2_cleanup);
if (!(line = get_uri_option_line(rdata->msg_info.to->uri))
&& !(line = get_uri_option_line(rdata->msg_info.msg->line.req.uri))) {
return NULL;
}
states = ao2_global_obj_ref(current_states);
if (!states) {
return NULL;
}
state = ao2_callback(states, 0, line_identify_relationship, line);
if (!state || ast_strlen_zero(state->registration->endpoint)) {
return NULL;
}
ast_debug(3, "Determined relationship to outbound registration '%s' based on line '%s', using configured endpoint '%s'\n",
ast_sorcery_object_get_id(state->registration), state->client_state->line, state->registration->endpoint);
return ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "endpoint", state->registration->endpoint);
}
static struct ast_sip_endpoint_identifier line_identifier = {
.identify_endpoint = line_identify,
};
/*! \brief Helper function which cancels the timer on a client */
static void cancel_registration(struct sip_outbound_registration_client_state *client_state)
{
if (pj_timer_heap_cancel_if_active(pjsip_endpt_get_timer_heap(ast_sip_get_pjsip_endpoint()),
&client_state->timer, client_state->timer.id)) {
/* The timer was successfully cancelled, drop the refcount of client_state */
ao2_ref(client_state, -1);
}
}
static pj_str_t PATH_NAME = { "path", 4 };
res_pjsip: Implement additional SIP RFCs for Google Voice trunk compatability This change implements a few different generic things which were brought on by Google Voice SIP. 1. The concept of flow transports have been introduced. These are configurable transports in pjsip.conf which can be used to reference a flow of signaling to a target. These have runtime configuration that can be changed by the signaling itself (such as Service-Routes and P-Preferred-Identity). When used these guarantee an individual connection (in the case of TCP or TLS) even if multiple flow transports exist to the same target. 2. Service-Routes (RFC 3608) support has been added to the outbound registration module which when received will be stored on the flow transport and used for requests referencing it. 3. P-Associated-URI / P-Preferred-Identity (RFC 3325) support has been added to the outbound registration module. If a P-Associated-URI header is received it will be used on requests as the P-Preferred-Identity. 4. Configurable outbound extension support has been added to the outbound registration module. When set the extension will be placed in the Supported header. 5. Header parameters can now be configured on an outbound registration which will be placed in the Contact header. 6. Google specific OAuth / Bearer token authentication (draft-ietf-sipcore-sip-authn-02) has been added to the outbound registration module. All functionality changes are controlled by pjsip.conf configuration options and do not affect non-configured pjsip endpoints otherwise. ASTERISK-27971 #close Change-Id: Id214c2d1c550a41fcf564b7df8f3da7be565bd58
2018-07-18 12:45:26 +00:00
static pj_str_t OUTBOUND_NAME = { "outbound", 8 };
AST_VECTOR(pjsip_generic_string_hdr_vector, pjsip_generic_string_hdr *);
/*!
* \internal
* \brief Callback function which finds a contact whose contact_status has security mechanisms.
*
* \param obj Pointer to the ast_sip_contact.
* \param arg Pointer-pointer to a contact_status that will be set to the contact_status found by this function.
* \param flags Flags used by the ao2_callback function.
*
* \note The refcount of the found contact_status must be decremented by the caller.
*/
static int contact_has_security_mechanisms(void *obj, void *arg, int flags)
{
struct ast_sip_contact *contact = obj;
struct ast_sip_contact_status **ret = arg;
struct ast_sip_contact_status *contact_status = ast_sip_get_contact_status(contact);
if (!contact_status) {
return -1;
}
if (!AST_VECTOR_SIZE(&contact_status->security_mechanisms)) {
ao2_cleanup(contact_status);
return -1;
}
*ret = contact_status;
return 0;
}
static int contact_add_security_headers_to_status(void *obj, void *arg, int flags)
{
struct ast_sip_contact *contact = obj;
struct pjsip_generic_string_hdr_vector *header_vector = arg;
struct ast_sip_contact_status *contact_status = ast_sip_get_contact_status(contact);
if (!contact_status) {
return -1;
}
if (AST_VECTOR_SIZE(&contact_status->security_mechanisms)) {
goto out;
}
ao2_lock(contact_status);
AST_VECTOR_CALLBACK_VOID(header_vector, ast_sip_header_to_security_mechanism, &contact_status->security_mechanisms);
ao2_unlock(contact_status);
out:
ao2_cleanup(contact_status);
return 0;
}
/*! \brief Adds security negotiation mechanisms of outbound registration client state as Security headers to tdata. */
static void add_security_headers(struct sip_outbound_registration_client_state *client_state,
pjsip_tx_data *tdata)
{
int add_require_header = 1;
int add_proxy_require_header = 1;
int add_sec_client_header = 0;
struct sip_outbound_registration *reg = NULL;
struct ast_sip_endpoint *endpt = NULL;
struct ao2_container *contact_container;
struct ast_sip_contact_status *contact_status = NULL;
struct ast_sip_security_mechanism_vector *sec_mechs = NULL;
static const pj_str_t security_verify = { "Security-Verify", 15 };
static const pj_str_t security_client = { "Security-Client", 15 };
static const pj_str_t proxy_require = { "Proxy-Require", 13 };
static const pj_str_t require = { "Require", 7 };
if (client_state->security_negotiation != AST_SIP_SECURITY_NEG_MEDIASEC) {
return;
}
/* Get contact status through registration -> endpoint name -> aor -> contact (if set) */
if ((reg = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "registration", client_state->registration_name))
&& !ast_strlen_zero(reg->endpoint) && (endpt = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "endpoint", reg->endpoint))
&& (contact_container = ast_sip_location_retrieve_contacts_from_aor_list(endpt->aors))) {
/* Retrieve all contacts associated with aors from this endpoint
* and find the first one that has security mechanisms.
*/
ao2_callback(contact_container, 0, contact_has_security_mechanisms, &contact_status);
if (contact_status) {
ao2_lock(contact_status);
sec_mechs = &contact_status->security_mechanisms;
}
ao2_cleanup(contact_container);
}
/* Use client_state->server_security_mechanisms if contact_status does not exist. */
if (!contact_status && AST_VECTOR_SIZE(&client_state->server_security_mechanisms)) {
sec_mechs = &client_state->server_security_mechanisms;
}
if (client_state->status == SIP_REGISTRATION_REJECTED_TEMPORARY || client_state->auth_attempted) {
if (sec_mechs != NULL && pjsip_msg_find_hdr_by_name(tdata->msg, &security_verify, NULL) == NULL) {
ast_sip_add_security_headers(sec_mechs, "Security-Verify", 0, tdata);
}
if (client_state->last_status_code == 494) {
ast_sip_remove_headers_by_name_and_value(tdata->msg, &security_client, NULL);
} else {
/* necessary if a retry occures */
add_sec_client_header = (pjsip_msg_find_hdr_by_name(tdata->msg, &security_client, NULL) == NULL) ? 1 : 0;
}
add_require_header =
(pjsip_msg_find_hdr_by_name(tdata->msg, &require, NULL) == NULL) ? 1 : 0;
add_proxy_require_header =
(pjsip_msg_find_hdr_by_name(tdata->msg, &proxy_require, NULL) == NULL) ? 1 : 0;
} else {
ast_sip_add_security_headers(&client_state->security_mechanisms, "Security-Client", 0, tdata);
}
if (add_require_header) {
ast_sip_add_header(tdata, "Require", "mediasec");
}
if (add_proxy_require_header) {
ast_sip_add_header(tdata, "Proxy-Require", "mediasec");
}
if (add_sec_client_header) {
ast_sip_add_security_headers(&client_state->security_mechanisms, "Security-Client", 0, tdata);
}
/* Cleanup */
if (contact_status) {
ao2_unlock(contact_status);
ao2_cleanup(contact_status);
}
ao2_cleanup(endpt);
ao2_cleanup(reg);
}
/*! \brief Helper function which sends a message and cleans up, if needed, on failure */
static pj_status_t registration_client_send(struct sip_outbound_registration_client_state *client_state,
pjsip_tx_data *tdata)
{
pj_status_t status;
int *callback_invoked;
pjsip_tpselector selector = { .type = PJSIP_TPSELECTOR_NONE, };
callback_invoked = ast_threadstorage_get(&register_callback_invoked, sizeof(int));
if (!callback_invoked) {
pjsip_tx_data_dec_ref(tdata);
return PJ_ENOMEM;
}
*callback_invoked = 0;
/* Due to the message going out the callback may now be invoked, so bump the count */
ao2_ref(client_state, +1);
res_pjsip_outbound_registration: Fix SRV failover on timeout In order to retry outbound registrations for some situations, we need access to the tdata from the original request. For instance, for 401/407 responses we need it to properly construct the subsequent request with the authentication. We also need it if we're iterating over a DNS SRV response record set so we can skip entries we've already tried. We've been getting the tdata from the server response rdata and transaction but that only works for the failures where there was actually a response (4XX, 5XX, etc). For timeouts there's no response and therefore no rdata or transaction from which to get the tdata. When processing a single A/AAAA record for a server, this wasn't an issue as we just retried that same server after the retry timer expired. If we got an SRV record set for the server though, without the state from the tdata, we just kept trying the first entry in the set repeatedly instead of skipping to the next one in the list. * Added a "last_tdata" member to the client state structure to keep track of the sent tdata. * Updated registration_client_send() to save the tdata it used into the client_state. * Updated sip_outbound_registration_response_cb() to use the tdata saved in client_state when we don't get a response from the server. We still use the tdata from the transaction when we DO get a response from the server so we can properly handle 4XX responses where our new request depends on it. General note on timeouts: Although res_pjsip_outbound_registration skips to the next record immediately when a timeout occurs during SRV set traversal, it's pjproject that determines how long to wait before a timeout is declared. As with other SIP message types, pjproject will continue trying the same server at an interval specified by "timer_t1" until "timer_b" expires. Both of those timers are set in the pjsip.conf "system" section. ASTERISK-28746 Change-Id: I199b8274392d17661dd3ce3b4d69a3968368fa06
2020-02-13 19:39:58 +00:00
/*
* We also bump tdata in expectation of saving it to client_state->last_tdata.
* We have to do it BEFORE pjsip_regc_send because if it succeeds, it decrements
* the ref count on its own.
*/
pjsip_tx_data_add_ref(tdata);
/* Add Security-Verify or Security-Client headers */
add_security_headers(client_state, tdata);
/*
* Replace the User-Agent header if a different one should be used
*/
if (!ast_strlen_zero(client_state->user_agent)) {
static const pj_str_t user_agent_str = { "User-Agent", 10 };
pjsip_generic_string_hdr *default_user_agent_hdr;
pjsip_generic_string_hdr *user_agent_hdr;
pj_str_t user_agent_val;
default_user_agent_hdr = pjsip_msg_find_hdr_by_name(tdata->msg, &user_agent_str, NULL);
user_agent_val = pj_str(client_state->user_agent);
user_agent_hdr = pjsip_generic_string_hdr_create(tdata->pool, &user_agent_str, &user_agent_val);
if (!user_agent_hdr) {
ast_log(LOG_ERROR, "Could not add custom User-Agent to outbound registration %s, sending REGISTER request with non-custom header\n", client_state->registration_name);
} else {
if (default_user_agent_hdr) {
pj_list_erase(default_user_agent_hdr);
}
pjsip_msg_add_hdr(tdata->msg, (pjsip_hdr *)user_agent_hdr);
}
}
/*
* Set the transport in case transports were reloaded.
* When pjproject removes the extraneous error messages produced,
* we can check status and only set the transport and resend if there was an error
*/
ast_sip_set_tpselector_from_transport_name(client_state->transport_name, &selector);
pjsip_regc_set_transport(client_state->client, &selector);
res_pjsip: Implement additional SIP RFCs for Google Voice trunk compatability This change implements a few different generic things which were brought on by Google Voice SIP. 1. The concept of flow transports have been introduced. These are configurable transports in pjsip.conf which can be used to reference a flow of signaling to a target. These have runtime configuration that can be changed by the signaling itself (such as Service-Routes and P-Preferred-Identity). When used these guarantee an individual connection (in the case of TCP or TLS) even if multiple flow transports exist to the same target. 2. Service-Routes (RFC 3608) support has been added to the outbound registration module which when received will be stored on the flow transport and used for requests referencing it. 3. P-Associated-URI / P-Preferred-Identity (RFC 3325) support has been added to the outbound registration module. If a P-Associated-URI header is received it will be used on requests as the P-Preferred-Identity. 4. Configurable outbound extension support has been added to the outbound registration module. When set the extension will be placed in the Supported header. 5. Header parameters can now be configured on an outbound registration which will be placed in the Contact header. 6. Google specific OAuth / Bearer token authentication (draft-ietf-sipcore-sip-authn-02) has been added to the outbound registration module. All functionality changes are controlled by pjsip.conf configuration options and do not affect non-configured pjsip endpoints otherwise. ASTERISK-27971 #close Change-Id: Id214c2d1c550a41fcf564b7df8f3da7be565bd58
2018-07-18 12:45:26 +00:00
ast_sip_tpselector_unref(&selector);
status = pjsip_regc_send(client_state->client, tdata);
res_pjsip_outbound_registration: Fix SRV failover on timeout In order to retry outbound registrations for some situations, we need access to the tdata from the original request. For instance, for 401/407 responses we need it to properly construct the subsequent request with the authentication. We also need it if we're iterating over a DNS SRV response record set so we can skip entries we've already tried. We've been getting the tdata from the server response rdata and transaction but that only works for the failures where there was actually a response (4XX, 5XX, etc). For timeouts there's no response and therefore no rdata or transaction from which to get the tdata. When processing a single A/AAAA record for a server, this wasn't an issue as we just retried that same server after the retry timer expired. If we got an SRV record set for the server though, without the state from the tdata, we just kept trying the first entry in the set repeatedly instead of skipping to the next one in the list. * Added a "last_tdata" member to the client state structure to keep track of the sent tdata. * Updated registration_client_send() to save the tdata it used into the client_state. * Updated sip_outbound_registration_response_cb() to use the tdata saved in client_state when we don't get a response from the server. We still use the tdata from the transaction when we DO get a response from the server so we can properly handle 4XX responses where our new request depends on it. General note on timeouts: Although res_pjsip_outbound_registration skips to the next record immediately when a timeout occurs during SRV set traversal, it's pjproject that determines how long to wait before a timeout is declared. As with other SIP message types, pjproject will continue trying the same server at an interval specified by "timer_t1" until "timer_b" expires. Both of those timers are set in the pjsip.conf "system" section. ASTERISK-28746 Change-Id: I199b8274392d17661dd3ce3b4d69a3968368fa06
2020-02-13 19:39:58 +00:00
/*
* If the attempt to send the message failed and the callback was not invoked we need to
* drop the references we just added
*/
if ((status != PJ_SUCCESS) && !(*callback_invoked)) {
res_pjsip_outbound_registration: Fix SRV failover on timeout In order to retry outbound registrations for some situations, we need access to the tdata from the original request. For instance, for 401/407 responses we need it to properly construct the subsequent request with the authentication. We also need it if we're iterating over a DNS SRV response record set so we can skip entries we've already tried. We've been getting the tdata from the server response rdata and transaction but that only works for the failures where there was actually a response (4XX, 5XX, etc). For timeouts there's no response and therefore no rdata or transaction from which to get the tdata. When processing a single A/AAAA record for a server, this wasn't an issue as we just retried that same server after the retry timer expired. If we got an SRV record set for the server though, without the state from the tdata, we just kept trying the first entry in the set repeatedly instead of skipping to the next one in the list. * Added a "last_tdata" member to the client state structure to keep track of the sent tdata. * Updated registration_client_send() to save the tdata it used into the client_state. * Updated sip_outbound_registration_response_cb() to use the tdata saved in client_state when we don't get a response from the server. We still use the tdata from the transaction when we DO get a response from the server so we can properly handle 4XX responses where our new request depends on it. General note on timeouts: Although res_pjsip_outbound_registration skips to the next record immediately when a timeout occurs during SRV set traversal, it's pjproject that determines how long to wait before a timeout is declared. As with other SIP message types, pjproject will continue trying the same server at an interval specified by "timer_t1" until "timer_b" expires. Both of those timers are set in the pjsip.conf "system" section. ASTERISK-28746 Change-Id: I199b8274392d17661dd3ce3b4d69a3968368fa06
2020-02-13 19:39:58 +00:00
pjsip_tx_data_dec_ref(tdata);
ao2_ref(client_state, -1);
res_pjsip_outbound_registration: Fix SRV failover on timeout In order to retry outbound registrations for some situations, we need access to the tdata from the original request. For instance, for 401/407 responses we need it to properly construct the subsequent request with the authentication. We also need it if we're iterating over a DNS SRV response record set so we can skip entries we've already tried. We've been getting the tdata from the server response rdata and transaction but that only works for the failures where there was actually a response (4XX, 5XX, etc). For timeouts there's no response and therefore no rdata or transaction from which to get the tdata. When processing a single A/AAAA record for a server, this wasn't an issue as we just retried that same server after the retry timer expired. If we got an SRV record set for the server though, without the state from the tdata, we just kept trying the first entry in the set repeatedly instead of skipping to the next one in the list. * Added a "last_tdata" member to the client state structure to keep track of the sent tdata. * Updated registration_client_send() to save the tdata it used into the client_state. * Updated sip_outbound_registration_response_cb() to use the tdata saved in client_state when we don't get a response from the server. We still use the tdata from the transaction when we DO get a response from the server so we can properly handle 4XX responses where our new request depends on it. General note on timeouts: Although res_pjsip_outbound_registration skips to the next record immediately when a timeout occurs during SRV set traversal, it's pjproject that determines how long to wait before a timeout is declared. As with other SIP message types, pjproject will continue trying the same server at an interval specified by "timer_t1" until "timer_b" expires. Both of those timers are set in the pjsip.conf "system" section. ASTERISK-28746 Change-Id: I199b8274392d17661dd3ce3b4d69a3968368fa06
2020-02-13 19:39:58 +00:00
return status;
}
res_pjsip_outbound_registration: Fix SRV failover on timeout In order to retry outbound registrations for some situations, we need access to the tdata from the original request. For instance, for 401/407 responses we need it to properly construct the subsequent request with the authentication. We also need it if we're iterating over a DNS SRV response record set so we can skip entries we've already tried. We've been getting the tdata from the server response rdata and transaction but that only works for the failures where there was actually a response (4XX, 5XX, etc). For timeouts there's no response and therefore no rdata or transaction from which to get the tdata. When processing a single A/AAAA record for a server, this wasn't an issue as we just retried that same server after the retry timer expired. If we got an SRV record set for the server though, without the state from the tdata, we just kept trying the first entry in the set repeatedly instead of skipping to the next one in the list. * Added a "last_tdata" member to the client state structure to keep track of the sent tdata. * Updated registration_client_send() to save the tdata it used into the client_state. * Updated sip_outbound_registration_response_cb() to use the tdata saved in client_state when we don't get a response from the server. We still use the tdata from the transaction when we DO get a response from the server so we can properly handle 4XX responses where our new request depends on it. General note on timeouts: Although res_pjsip_outbound_registration skips to the next record immediately when a timeout occurs during SRV set traversal, it's pjproject that determines how long to wait before a timeout is declared. As with other SIP message types, pjproject will continue trying the same server at an interval specified by "timer_t1" until "timer_b" expires. Both of those timers are set in the pjsip.conf "system" section. ASTERISK-28746 Change-Id: I199b8274392d17661dd3ce3b4d69a3968368fa06
2020-02-13 19:39:58 +00:00
/*
* Decref the old last_data before replacing it.
* BTW, it's quite possible that last_data == tdata
* if we're trying successive servers in an SRV set.
*/
if (client_state->last_tdata) {
pjsip_tx_data_dec_ref(client_state->last_tdata);
}
client_state->last_tdata = tdata;
return status;
}
res_pjsip: Implement additional SIP RFCs for Google Voice trunk compatability This change implements a few different generic things which were brought on by Google Voice SIP. 1. The concept of flow transports have been introduced. These are configurable transports in pjsip.conf which can be used to reference a flow of signaling to a target. These have runtime configuration that can be changed by the signaling itself (such as Service-Routes and P-Preferred-Identity). When used these guarantee an individual connection (in the case of TCP or TLS) even if multiple flow transports exist to the same target. 2. Service-Routes (RFC 3608) support has been added to the outbound registration module which when received will be stored on the flow transport and used for requests referencing it. 3. P-Associated-URI / P-Preferred-Identity (RFC 3325) support has been added to the outbound registration module. If a P-Associated-URI header is received it will be used on requests as the P-Preferred-Identity. 4. Configurable outbound extension support has been added to the outbound registration module. When set the extension will be placed in the Supported header. 5. Header parameters can now be configured on an outbound registration which will be placed in the Contact header. 6. Google specific OAuth / Bearer token authentication (draft-ietf-sipcore-sip-authn-02) has been added to the outbound registration module. All functionality changes are controlled by pjsip.conf configuration options and do not affect non-configured pjsip endpoints otherwise. ASTERISK-27971 #close Change-Id: Id214c2d1c550a41fcf564b7df8f3da7be565bd58
2018-07-18 12:45:26 +00:00
/*! \brief Helper function to add string to Supported header */
static int add_to_supported_header(pjsip_tx_data *tdata, pj_str_t *name)
{
pjsip_supported_hdr *hdr;
int i;
res_pjsip: Implement additional SIP RFCs for Google Voice trunk compatability This change implements a few different generic things which were brought on by Google Voice SIP. 1. The concept of flow transports have been introduced. These are configurable transports in pjsip.conf which can be used to reference a flow of signaling to a target. These have runtime configuration that can be changed by the signaling itself (such as Service-Routes and P-Preferred-Identity). When used these guarantee an individual connection (in the case of TCP or TLS) even if multiple flow transports exist to the same target. 2. Service-Routes (RFC 3608) support has been added to the outbound registration module which when received will be stored on the flow transport and used for requests referencing it. 3. P-Associated-URI / P-Preferred-Identity (RFC 3325) support has been added to the outbound registration module. If a P-Associated-URI header is received it will be used on requests as the P-Preferred-Identity. 4. Configurable outbound extension support has been added to the outbound registration module. When set the extension will be placed in the Supported header. 5. Header parameters can now be configured on an outbound registration which will be placed in the Contact header. 6. Google specific OAuth / Bearer token authentication (draft-ietf-sipcore-sip-authn-02) has been added to the outbound registration module. All functionality changes are controlled by pjsip.conf configuration options and do not affect non-configured pjsip endpoints otherwise. ASTERISK-27971 #close Change-Id: Id214c2d1c550a41fcf564b7df8f3da7be565bd58
2018-07-18 12:45:26 +00:00
hdr = pjsip_msg_find_hdr(tdata->msg, PJSIP_H_SUPPORTED, NULL);
if (!hdr) {
/* insert a new Supported header */
hdr = pjsip_supported_hdr_create(tdata->pool);
if (!hdr) {
pjsip_tx_data_dec_ref(tdata);
return 0;
}
pjsip_msg_add_hdr(tdata->msg, (pjsip_hdr *)hdr);
}
/* Don't add the value if it's already there */
for (i = 0; i < hdr->count; ++i) {
if (pj_stricmp(&hdr->values[i], name) == 0) {
return 1;
}
}
if (hdr->count >= PJSIP_GENERIC_ARRAY_MAX_COUNT) {
return 0;
}
res_pjsip: Implement additional SIP RFCs for Google Voice trunk compatability This change implements a few different generic things which were brought on by Google Voice SIP. 1. The concept of flow transports have been introduced. These are configurable transports in pjsip.conf which can be used to reference a flow of signaling to a target. These have runtime configuration that can be changed by the signaling itself (such as Service-Routes and P-Preferred-Identity). When used these guarantee an individual connection (in the case of TCP or TLS) even if multiple flow transports exist to the same target. 2. Service-Routes (RFC 3608) support has been added to the outbound registration module which when received will be stored on the flow transport and used for requests referencing it. 3. P-Associated-URI / P-Preferred-Identity (RFC 3325) support has been added to the outbound registration module. If a P-Associated-URI header is received it will be used on requests as the P-Preferred-Identity. 4. Configurable outbound extension support has been added to the outbound registration module. When set the extension will be placed in the Supported header. 5. Header parameters can now be configured on an outbound registration which will be placed in the Contact header. 6. Google specific OAuth / Bearer token authentication (draft-ietf-sipcore-sip-authn-02) has been added to the outbound registration module. All functionality changes are controlled by pjsip.conf configuration options and do not affect non-configured pjsip endpoints otherwise. ASTERISK-27971 #close Change-Id: Id214c2d1c550a41fcf564b7df8f3da7be565bd58
2018-07-18 12:45:26 +00:00
/* add on to the existing Supported header */
pj_strassign(&hdr->values[hdr->count++], name);
return 1;
}
/*! \brief Helper function to add configured supported headers */
static int add_configured_supported_headers(struct sip_outbound_registration_client_state *client_state, pjsip_tx_data *tdata)
{
if (client_state->support_path) {
if (!add_to_supported_header(tdata, &PATH_NAME)) {
return 0;
}
}
if (client_state->support_outbound) {
if (!add_to_supported_header(tdata, &OUTBOUND_NAME)) {
return 0;
}
}
return 1;
}
/*! \brief Callback function for registering */
static int handle_client_registration(void *data)
{
RAII_VAR(struct sip_outbound_registration_client_state *, client_state, data, ao2_cleanup);
pjsip_tx_data *tdata;
res_pjsip: Implement additional SIP RFCs for Google Voice trunk compatability This change implements a few different generic things which were brought on by Google Voice SIP. 1. The concept of flow transports have been introduced. These are configurable transports in pjsip.conf which can be used to reference a flow of signaling to a target. These have runtime configuration that can be changed by the signaling itself (such as Service-Routes and P-Preferred-Identity). When used these guarantee an individual connection (in the case of TCP or TLS) even if multiple flow transports exist to the same target. 2. Service-Routes (RFC 3608) support has been added to the outbound registration module which when received will be stored on the flow transport and used for requests referencing it. 3. P-Associated-URI / P-Preferred-Identity (RFC 3325) support has been added to the outbound registration module. If a P-Associated-URI header is received it will be used on requests as the P-Preferred-Identity. 4. Configurable outbound extension support has been added to the outbound registration module. When set the extension will be placed in the Supported header. 5. Header parameters can now be configured on an outbound registration which will be placed in the Contact header. 6. Google specific OAuth / Bearer token authentication (draft-ietf-sipcore-sip-authn-02) has been added to the outbound registration module. All functionality changes are controlled by pjsip.conf configuration options and do not affect non-configured pjsip endpoints otherwise. ASTERISK-27971 #close Change-Id: Id214c2d1c550a41fcf564b7df8f3da7be565bd58
2018-07-18 12:45:26 +00:00
if (set_outbound_initial_authentication_credentials(client_state->client, &client_state->outbound_auths)) {
ast_log(LOG_WARNING, "Failed to set initial authentication credentials\n");
}
if (client_state->status == SIP_REGISTRATION_STOPPED
|| pjsip_regc_register(client_state->client, PJ_FALSE, &tdata) != PJ_SUCCESS) {
return 0;
}
if (DEBUG_ATLEAST(1)) {
pjsip_regc_info info;
pjsip_regc_get_info(client_state->client, &info);
ast_log(LOG_DEBUG, "Outbound REGISTER attempt %u to '%.*s' with client '%.*s'\n",
client_state->retries + 1,
(int) info.server_uri.slen, info.server_uri.ptr,
(int) info.client_uri.slen, info.client_uri.ptr);
}
res_pjsip: Implement additional SIP RFCs for Google Voice trunk compatability This change implements a few different generic things which were brought on by Google Voice SIP. 1. The concept of flow transports have been introduced. These are configurable transports in pjsip.conf which can be used to reference a flow of signaling to a target. These have runtime configuration that can be changed by the signaling itself (such as Service-Routes and P-Preferred-Identity). When used these guarantee an individual connection (in the case of TCP or TLS) even if multiple flow transports exist to the same target. 2. Service-Routes (RFC 3608) support has been added to the outbound registration module which when received will be stored on the flow transport and used for requests referencing it. 3. P-Associated-URI / P-Preferred-Identity (RFC 3325) support has been added to the outbound registration module. If a P-Associated-URI header is received it will be used on requests as the P-Preferred-Identity. 4. Configurable outbound extension support has been added to the outbound registration module. When set the extension will be placed in the Supported header. 5. Header parameters can now be configured on an outbound registration which will be placed in the Contact header. 6. Google specific OAuth / Bearer token authentication (draft-ietf-sipcore-sip-authn-02) has been added to the outbound registration module. All functionality changes are controlled by pjsip.conf configuration options and do not affect non-configured pjsip endpoints otherwise. ASTERISK-27971 #close Change-Id: Id214c2d1c550a41fcf564b7df8f3da7be565bd58
2018-07-18 12:45:26 +00:00
if (!add_configured_supported_headers(client_state, tdata)) {
ast_log(LOG_WARNING, "Failed to set supported headers\n");
return -1;
}
registration_client_send(client_state, tdata);
return 0;
}
/*! \brief Timer callback function, used just for registrations */
static void sip_outbound_registration_timer_cb(pj_timer_heap_t *timer_heap, struct pj_timer_entry *entry)
{
struct sip_outbound_registration_client_state *client_state = entry->user_data;
entry->id = 0;
/*
* Transfer client_state reference to serializer task so the
* nominal path will not dec the client_state ref in this
* pjproject callback thread.
*/
if (ast_sip_push_task(client_state->serializer, handle_client_registration, client_state)) {
ast_log(LOG_WARNING, "Scheduled outbound registration could not be executed.\n");
ao2_ref(client_state, -1);
}
}
/*! \brief Helper function which sets up the timer to re-register in a specific amount of time */
static void schedule_registration(struct sip_outbound_registration_client_state *client_state, unsigned int seconds)
{
pj_time_val delay = { .sec = seconds, };
pjsip_regc_info info;
cancel_registration(client_state);
pjsip_regc_get_info(client_state->client, &info);
ast_debug(1, "Scheduling outbound registration to server '%.*s' from client '%.*s' in %d seconds\n",
(int) info.server_uri.slen, info.server_uri.ptr,
(int) info.client_uri.slen, info.client_uri.ptr,
seconds);
ao2_ref(client_state, +1);
if (pjsip_endpt_schedule_timer(ast_sip_get_pjsip_endpoint(), &client_state->timer, &delay) != PJ_SUCCESS) {
ast_log(LOG_WARNING, "Failed to schedule registration to server '%.*s' from client '%.*s'\n",
(int) info.server_uri.slen, info.server_uri.ptr,
(int) info.client_uri.slen, info.client_uri.ptr);
ao2_ref(client_state, -1);
}
client_state->registration_expires = ((int) time(NULL)) + seconds;
}
static void update_client_state_status(struct sip_outbound_registration_client_state *client_state, enum sip_outbound_registration_status status)
{
const char *status_old;
const char *status_new;
if (client_state->status == status) {
/* Status state did not change at all. */
return;
}
status_old = sip_outbound_registration_status_str(client_state->status);
status_new = sip_outbound_registration_status_str(status);
client_state->status = status;
if (!strcmp(status_old, status_new)) {
/*
* The internal status state may have changed but the status
* state we tell the world did not change at all.
*/
return;
}
ast_statsd_log_string_va("PJSIP.registrations.state.%s", AST_STATSD_GAUGE, "-1", 1.0,
status_old);
ast_statsd_log_string_va("PJSIP.registrations.state.%s", AST_STATSD_GAUGE, "+1", 1.0,
status_new);
}
/*! \brief Callback function for unregistering (potentially) and destroying state */
static int handle_client_state_destruction(void *data)
{
res_pjsip_outbound_registration.c: Fix handle_client_state_destruction() refs * handle_client_state_destruction() must always be passed a ref to client_state because it will always unref client_state. handle_registration_response() was not passing a client_state ref. * Made the final un-REGISTER message get sent normally using the pjproject register control structure in handle_client_state_destruction(). The previous code attempted to short circuit the response handling for the module to unload. That doesn't work for a couple reasons. One, pjsip_regc_send() may call the registered callback before it returns and unbalance the client_state ref count. Two, the registered callback handles any authentication for the un-REGISTER message. * Made the distinction between internal registration state and external registration status with sip_outbound_registration_status_str(). This is necessary to avoid altering documented AMI messages with internal changes. * Removed references to client_state->client outside of the serializer thread. When handle_client_state_destruction() destroys the pjproject register control structure that memory is freed and cannot be referenced anymore. These accesses were to provide information for debug and off-nominal warning messages. * In sip_outbound_registration_timer_cb() you should not access entry->id after unrefing client_state because the passed in entry is normally pointing to the timer entry in the client_state object. ASTERISK-24907 Reported by: Kevin Harwell Change-Id: Ia7b446d8644b6b4550ef5bea49527671de65183f
2015-06-16 20:06:22 +00:00
struct sip_outbound_registration_client_state *client_state = data;
cancel_registration(client_state);
if (client_state->client) {
pjsip_regc_info info;
res_pjsip_outbound_registration.c: Fix handle_client_state_destruction() refs * handle_client_state_destruction() must always be passed a ref to client_state because it will always unref client_state. handle_registration_response() was not passing a client_state ref. * Made the final un-REGISTER message get sent normally using the pjproject register control structure in handle_client_state_destruction(). The previous code attempted to short circuit the response handling for the module to unload. That doesn't work for a couple reasons. One, pjsip_regc_send() may call the registered callback before it returns and unbalance the client_state ref count. Two, the registered callback handles any authentication for the un-REGISTER message. * Made the distinction between internal registration state and external registration status with sip_outbound_registration_status_str(). This is necessary to avoid altering documented AMI messages with internal changes. * Removed references to client_state->client outside of the serializer thread. When handle_client_state_destruction() destroys the pjproject register control structure that memory is freed and cannot be referenced anymore. These accesses were to provide information for debug and off-nominal warning messages. * In sip_outbound_registration_timer_cb() you should not access entry->id after unrefing client_state because the passed in entry is normally pointing to the timer entry in the client_state object. ASTERISK-24907 Reported by: Kevin Harwell Change-Id: Ia7b446d8644b6b4550ef5bea49527671de65183f
2015-06-16 20:06:22 +00:00
pjsip_tx_data *tdata;
pjsip_regc_get_info(client_state->client, &info);
if (info.is_busy == PJ_TRUE) {
/* If a client transaction is in progress we defer until it is complete */
res_pjsip_outbound_registration.c: Fix handle_client_state_destruction() refs * handle_client_state_destruction() must always be passed a ref to client_state because it will always unref client_state. handle_registration_response() was not passing a client_state ref. * Made the final un-REGISTER message get sent normally using the pjproject register control structure in handle_client_state_destruction(). The previous code attempted to short circuit the response handling for the module to unload. That doesn't work for a couple reasons. One, pjsip_regc_send() may call the registered callback before it returns and unbalance the client_state ref count. Two, the registered callback handles any authentication for the un-REGISTER message. * Made the distinction between internal registration state and external registration status with sip_outbound_registration_status_str(). This is necessary to avoid altering documented AMI messages with internal changes. * Removed references to client_state->client outside of the serializer thread. When handle_client_state_destruction() destroys the pjproject register control structure that memory is freed and cannot be referenced anymore. These accesses were to provide information for debug and off-nominal warning messages. * In sip_outbound_registration_timer_cb() you should not access entry->id after unrefing client_state because the passed in entry is normally pointing to the timer entry in the client_state object. ASTERISK-24907 Reported by: Kevin Harwell Change-Id: Ia7b446d8644b6b4550ef5bea49527671de65183f
2015-06-16 20:06:22 +00:00
ast_debug(1,
"Registration transaction is busy with server '%.*s' from client '%.*s'.\n",
res_pjsip_outbound_registration.c: Fix handle_client_state_destruction() refs * handle_client_state_destruction() must always be passed a ref to client_state because it will always unref client_state. handle_registration_response() was not passing a client_state ref. * Made the final un-REGISTER message get sent normally using the pjproject register control structure in handle_client_state_destruction(). The previous code attempted to short circuit the response handling for the module to unload. That doesn't work for a couple reasons. One, pjsip_regc_send() may call the registered callback before it returns and unbalance the client_state ref count. Two, the registered callback handles any authentication for the un-REGISTER message. * Made the distinction between internal registration state and external registration status with sip_outbound_registration_status_str(). This is necessary to avoid altering documented AMI messages with internal changes. * Removed references to client_state->client outside of the serializer thread. When handle_client_state_destruction() destroys the pjproject register control structure that memory is freed and cannot be referenced anymore. These accesses were to provide information for debug and off-nominal warning messages. * In sip_outbound_registration_timer_cb() you should not access entry->id after unrefing client_state because the passed in entry is normally pointing to the timer entry in the client_state object. ASTERISK-24907 Reported by: Kevin Harwell Change-Id: Ia7b446d8644b6b4550ef5bea49527671de65183f
2015-06-16 20:06:22 +00:00
(int) info.server_uri.slen, info.server_uri.ptr,
(int) info.client_uri.slen, info.client_uri.ptr);
client_state->destroy = 1;
res_pjsip_outbound_registration.c: Fix handle_client_state_destruction() refs * handle_client_state_destruction() must always be passed a ref to client_state because it will always unref client_state. handle_registration_response() was not passing a client_state ref. * Made the final un-REGISTER message get sent normally using the pjproject register control structure in handle_client_state_destruction(). The previous code attempted to short circuit the response handling for the module to unload. That doesn't work for a couple reasons. One, pjsip_regc_send() may call the registered callback before it returns and unbalance the client_state ref count. Two, the registered callback handles any authentication for the un-REGISTER message. * Made the distinction between internal registration state and external registration status with sip_outbound_registration_status_str(). This is necessary to avoid altering documented AMI messages with internal changes. * Removed references to client_state->client outside of the serializer thread. When handle_client_state_destruction() destroys the pjproject register control structure that memory is freed and cannot be referenced anymore. These accesses were to provide information for debug and off-nominal warning messages. * In sip_outbound_registration_timer_cb() you should not access entry->id after unrefing client_state because the passed in entry is normally pointing to the timer entry in the client_state object. ASTERISK-24907 Reported by: Kevin Harwell Change-Id: Ia7b446d8644b6b4550ef5bea49527671de65183f
2015-06-16 20:06:22 +00:00
ao2_ref(client_state, -1);
return 0;
}
res_pjsip_outbound_registration.c: Fix handle_client_state_destruction() refs * handle_client_state_destruction() must always be passed a ref to client_state because it will always unref client_state. handle_registration_response() was not passing a client_state ref. * Made the final un-REGISTER message get sent normally using the pjproject register control structure in handle_client_state_destruction(). The previous code attempted to short circuit the response handling for the module to unload. That doesn't work for a couple reasons. One, pjsip_regc_send() may call the registered callback before it returns and unbalance the client_state ref count. Two, the registered callback handles any authentication for the un-REGISTER message. * Made the distinction between internal registration state and external registration status with sip_outbound_registration_status_str(). This is necessary to avoid altering documented AMI messages with internal changes. * Removed references to client_state->client outside of the serializer thread. When handle_client_state_destruction() destroys the pjproject register control structure that memory is freed and cannot be referenced anymore. These accesses were to provide information for debug and off-nominal warning messages. * In sip_outbound_registration_timer_cb() you should not access entry->id after unrefing client_state because the passed in entry is normally pointing to the timer entry in the client_state object. ASTERISK-24907 Reported by: Kevin Harwell Change-Id: Ia7b446d8644b6b4550ef5bea49527671de65183f
2015-06-16 20:06:22 +00:00
switch (client_state->status) {
case SIP_REGISTRATION_UNREGISTERED:
break;
case SIP_REGISTRATION_REGISTERED:
ast_debug(1,
"Trying to unregister with server '%.*s' from client '%.*s' before destruction.\n",
res_pjsip_outbound_registration.c: Fix handle_client_state_destruction() refs * handle_client_state_destruction() must always be passed a ref to client_state because it will always unref client_state. handle_registration_response() was not passing a client_state ref. * Made the final un-REGISTER message get sent normally using the pjproject register control structure in handle_client_state_destruction(). The previous code attempted to short circuit the response handling for the module to unload. That doesn't work for a couple reasons. One, pjsip_regc_send() may call the registered callback before it returns and unbalance the client_state ref count. Two, the registered callback handles any authentication for the un-REGISTER message. * Made the distinction between internal registration state and external registration status with sip_outbound_registration_status_str(). This is necessary to avoid altering documented AMI messages with internal changes. * Removed references to client_state->client outside of the serializer thread. When handle_client_state_destruction() destroys the pjproject register control structure that memory is freed and cannot be referenced anymore. These accesses were to provide information for debug and off-nominal warning messages. * In sip_outbound_registration_timer_cb() you should not access entry->id after unrefing client_state because the passed in entry is normally pointing to the timer entry in the client_state object. ASTERISK-24907 Reported by: Kevin Harwell Change-Id: Ia7b446d8644b6b4550ef5bea49527671de65183f
2015-06-16 20:06:22 +00:00
(int) info.server_uri.slen, info.server_uri.ptr,
(int) info.client_uri.slen, info.client_uri.ptr);
update_client_state_status(client_state, SIP_REGISTRATION_STOPPING);
res_pjsip_outbound_registration.c: Fix handle_client_state_destruction() refs * handle_client_state_destruction() must always be passed a ref to client_state because it will always unref client_state. handle_registration_response() was not passing a client_state ref. * Made the final un-REGISTER message get sent normally using the pjproject register control structure in handle_client_state_destruction(). The previous code attempted to short circuit the response handling for the module to unload. That doesn't work for a couple reasons. One, pjsip_regc_send() may call the registered callback before it returns and unbalance the client_state ref count. Two, the registered callback handles any authentication for the un-REGISTER message. * Made the distinction between internal registration state and external registration status with sip_outbound_registration_status_str(). This is necessary to avoid altering documented AMI messages with internal changes. * Removed references to client_state->client outside of the serializer thread. When handle_client_state_destruction() destroys the pjproject register control structure that memory is freed and cannot be referenced anymore. These accesses were to provide information for debug and off-nominal warning messages. * In sip_outbound_registration_timer_cb() you should not access entry->id after unrefing client_state because the passed in entry is normally pointing to the timer entry in the client_state object. ASTERISK-24907 Reported by: Kevin Harwell Change-Id: Ia7b446d8644b6b4550ef5bea49527671de65183f
2015-06-16 20:06:22 +00:00
client_state->destroy = 1;
if (pjsip_regc_unregister(client_state->client, &tdata) == PJ_SUCCESS
res_pjsip: Implement additional SIP RFCs for Google Voice trunk compatability This change implements a few different generic things which were brought on by Google Voice SIP. 1. The concept of flow transports have been introduced. These are configurable transports in pjsip.conf which can be used to reference a flow of signaling to a target. These have runtime configuration that can be changed by the signaling itself (such as Service-Routes and P-Preferred-Identity). When used these guarantee an individual connection (in the case of TCP or TLS) even if multiple flow transports exist to the same target. 2. Service-Routes (RFC 3608) support has been added to the outbound registration module which when received will be stored on the flow transport and used for requests referencing it. 3. P-Associated-URI / P-Preferred-Identity (RFC 3325) support has been added to the outbound registration module. If a P-Associated-URI header is received it will be used on requests as the P-Preferred-Identity. 4. Configurable outbound extension support has been added to the outbound registration module. When set the extension will be placed in the Supported header. 5. Header parameters can now be configured on an outbound registration which will be placed in the Contact header. 6. Google specific OAuth / Bearer token authentication (draft-ietf-sipcore-sip-authn-02) has been added to the outbound registration module. All functionality changes are controlled by pjsip.conf configuration options and do not affect non-configured pjsip endpoints otherwise. ASTERISK-27971 #close Change-Id: Id214c2d1c550a41fcf564b7df8f3da7be565bd58
2018-07-18 12:45:26 +00:00
&& add_configured_supported_headers(client_state, tdata)
res_pjsip_outbound_registration.c: Fix handle_client_state_destruction() refs * handle_client_state_destruction() must always be passed a ref to client_state because it will always unref client_state. handle_registration_response() was not passing a client_state ref. * Made the final un-REGISTER message get sent normally using the pjproject register control structure in handle_client_state_destruction(). The previous code attempted to short circuit the response handling for the module to unload. That doesn't work for a couple reasons. One, pjsip_regc_send() may call the registered callback before it returns and unbalance the client_state ref count. Two, the registered callback handles any authentication for the un-REGISTER message. * Made the distinction between internal registration state and external registration status with sip_outbound_registration_status_str(). This is necessary to avoid altering documented AMI messages with internal changes. * Removed references to client_state->client outside of the serializer thread. When handle_client_state_destruction() destroys the pjproject register control structure that memory is freed and cannot be referenced anymore. These accesses were to provide information for debug and off-nominal warning messages. * In sip_outbound_registration_timer_cb() you should not access entry->id after unrefing client_state because the passed in entry is normally pointing to the timer entry in the client_state object. ASTERISK-24907 Reported by: Kevin Harwell Change-Id: Ia7b446d8644b6b4550ef5bea49527671de65183f
2015-06-16 20:06:22 +00:00
&& registration_client_send(client_state, tdata) == PJ_SUCCESS) {
ao2_ref(client_state, -1);
return 0;
}
res_pjsip_outbound_registration.c: Fix handle_client_state_destruction() refs * handle_client_state_destruction() must always be passed a ref to client_state because it will always unref client_state. handle_registration_response() was not passing a client_state ref. * Made the final un-REGISTER message get sent normally using the pjproject register control structure in handle_client_state_destruction(). The previous code attempted to short circuit the response handling for the module to unload. That doesn't work for a couple reasons. One, pjsip_regc_send() may call the registered callback before it returns and unbalance the client_state ref count. Two, the registered callback handles any authentication for the un-REGISTER message. * Made the distinction between internal registration state and external registration status with sip_outbound_registration_status_str(). This is necessary to avoid altering documented AMI messages with internal changes. * Removed references to client_state->client outside of the serializer thread. When handle_client_state_destruction() destroys the pjproject register control structure that memory is freed and cannot be referenced anymore. These accesses were to provide information for debug and off-nominal warning messages. * In sip_outbound_registration_timer_cb() you should not access entry->id after unrefing client_state because the passed in entry is normally pointing to the timer entry in the client_state object. ASTERISK-24907 Reported by: Kevin Harwell Change-Id: Ia7b446d8644b6b4550ef5bea49527671de65183f
2015-06-16 20:06:22 +00:00
break;
case SIP_REGISTRATION_REJECTED_TEMPORARY:
case SIP_REGISTRATION_REJECTED_PERMANENT:
case SIP_REGISTRATION_STOPPING:
res_pjsip_outbound_registration.c: Fix handle_client_state_destruction() refs * handle_client_state_destruction() must always be passed a ref to client_state because it will always unref client_state. handle_registration_response() was not passing a client_state ref. * Made the final un-REGISTER message get sent normally using the pjproject register control structure in handle_client_state_destruction(). The previous code attempted to short circuit the response handling for the module to unload. That doesn't work for a couple reasons. One, pjsip_regc_send() may call the registered callback before it returns and unbalance the client_state ref count. Two, the registered callback handles any authentication for the un-REGISTER message. * Made the distinction between internal registration state and external registration status with sip_outbound_registration_status_str(). This is necessary to avoid altering documented AMI messages with internal changes. * Removed references to client_state->client outside of the serializer thread. When handle_client_state_destruction() destroys the pjproject register control structure that memory is freed and cannot be referenced anymore. These accesses were to provide information for debug and off-nominal warning messages. * In sip_outbound_registration_timer_cb() you should not access entry->id after unrefing client_state because the passed in entry is normally pointing to the timer entry in the client_state object. ASTERISK-24907 Reported by: Kevin Harwell Change-Id: Ia7b446d8644b6b4550ef5bea49527671de65183f
2015-06-16 20:06:22 +00:00
case SIP_REGISTRATION_STOPPED:
break;
}
pjsip_regc_destroy(client_state->client);
res_pjsip_outbound_registration.c: Fix handle_client_state_destruction() refs * handle_client_state_destruction() must always be passed a ref to client_state because it will always unref client_state. handle_registration_response() was not passing a client_state ref. * Made the final un-REGISTER message get sent normally using the pjproject register control structure in handle_client_state_destruction(). The previous code attempted to short circuit the response handling for the module to unload. That doesn't work for a couple reasons. One, pjsip_regc_send() may call the registered callback before it returns and unbalance the client_state ref count. Two, the registered callback handles any authentication for the un-REGISTER message. * Made the distinction between internal registration state and external registration status with sip_outbound_registration_status_str(). This is necessary to avoid altering documented AMI messages with internal changes. * Removed references to client_state->client outside of the serializer thread. When handle_client_state_destruction() destroys the pjproject register control structure that memory is freed and cannot be referenced anymore. These accesses were to provide information for debug and off-nominal warning messages. * In sip_outbound_registration_timer_cb() you should not access entry->id after unrefing client_state because the passed in entry is normally pointing to the timer entry in the client_state object. ASTERISK-24907 Reported by: Kevin Harwell Change-Id: Ia7b446d8644b6b4550ef5bea49527671de65183f
2015-06-16 20:06:22 +00:00
client_state->client = NULL;
}
update_client_state_status(client_state, SIP_REGISTRATION_STOPPED);
ast_sip_auth_vector_destroy(&client_state->outbound_auths);
ast_sip_security_mechanisms_vector_destroy(&client_state->security_mechanisms);
ast_sip_security_mechanisms_vector_destroy(&client_state->server_security_mechanisms);
res_pjsip_outbound_registration.c: Fix handle_client_state_destruction() refs * handle_client_state_destruction() must always be passed a ref to client_state because it will always unref client_state. handle_registration_response() was not passing a client_state ref. * Made the final un-REGISTER message get sent normally using the pjproject register control structure in handle_client_state_destruction(). The previous code attempted to short circuit the response handling for the module to unload. That doesn't work for a couple reasons. One, pjsip_regc_send() may call the registered callback before it returns and unbalance the client_state ref count. Two, the registered callback handles any authentication for the un-REGISTER message. * Made the distinction between internal registration state and external registration status with sip_outbound_registration_status_str(). This is necessary to avoid altering documented AMI messages with internal changes. * Removed references to client_state->client outside of the serializer thread. When handle_client_state_destruction() destroys the pjproject register control structure that memory is freed and cannot be referenced anymore. These accesses were to provide information for debug and off-nominal warning messages. * In sip_outbound_registration_timer_cb() you should not access entry->id after unrefing client_state because the passed in entry is normally pointing to the timer entry in the client_state object. ASTERISK-24907 Reported by: Kevin Harwell Change-Id: Ia7b446d8644b6b4550ef5bea49527671de65183f
2015-06-16 20:06:22 +00:00
ao2_ref(client_state, -1);
return 0;
}
/*! \brief Structure for registration response */
struct registration_response {
/*! \brief Response code for the registration attempt */
int code;
/*! \brief Expiration time for registration */
int expiration;
/*! \brief Retry-After value */
int retry_after;
/*! \brief Outbound registration client state */
struct sip_outbound_registration_client_state *client_state;
/*! \brief The response message */
pjsip_rx_data *rdata;
/*! \brief Request for which the response was received */
pjsip_tx_data *old_request;
pjsip_transport_events: Fix possible use after free on transport It was possible for a module that registered for transport monitor events to pass in a pjsip_transport that had already been freed. This caused pjsip_transport_events to crash when looking up the monitor for the transport. The fix is a two pronged approach. 1. We now increment the reference count on pjsip_transports when we create monitors for them, then decrement the count when the transport is going to be destroyed. 2. There are now APIs to register and unregister monitor callbacks by "transport key" which is a string concatenation of the remote ip address and port. This way the module needing to monitor the transport doesn't have to hold on to the transport object itself to unregister. It just has to save the transport_key. * Added the pjsip_transport reference increment and decrement. * Changed the internal transport monitor container key from the transport->obj_name (which may not be unique anyway) to the transport_key. * Added a helper macro AST_SIP_MAKE_REMOTE_IPADDR_PORT_STR() that fills a buffer with the transport_key using a passed-in pjsip_transport. * Added the following functions: ast_sip_transport_monitor_register_key ast_sip_transport_monitor_register_replace_key ast_sip_transport_monitor_unregister_key and marked their non-key counterparts as deprecated. * Updated res_pjsip_pubsub and res_pjsip_outbound_register to use the new "key" monitor functions. NOTE: res_pjsip_registrar also uses the transport monitor functionality but doesn't have a persistent object other than contact to store a transport key. At this time, it continues to use the non-key monitor functions. ASTERISK-30244 Change-Id: I1a20baf2a8643c272dcf819871d6c395f148f00b (cherry picked from commit 7684c9e907fb85f5c58b025d9e385ad2600f12a2)
2022-10-10 14:35:54 +00:00
/*! \brief Key for the reliable transport in use */
char transport_key[IP6ADDR_COLON_PORT_BUFLEN];
};
/*! \brief Registration response structure destructor */
static void registration_response_destroy(void *obj)
{
struct registration_response *response = obj;
if (response->rdata) {
pjsip_rx_data_free_cloned(response->rdata);
}
if (response->old_request) {
pjsip_tx_data_dec_ref(response->old_request);
}
ao2_cleanup(response->client_state);
}
/*! \brief Helper function which determines if a response code is temporal or not */
static int sip_outbound_registration_is_temporal(unsigned int code,
struct sip_outbound_registration_client_state *client_state)
{
/* Shamelessly taken from pjsua */
if (code == PJSIP_SC_REQUEST_TIMEOUT ||
code == PJSIP_SC_INTERNAL_SERVER_ERROR ||
code == PJSIP_SC_BAD_GATEWAY ||
code == PJSIP_SC_SERVICE_UNAVAILABLE ||
code == PJSIP_SC_SERVER_TIMEOUT ||
((code == PJSIP_SC_UNAUTHORIZED ||
code == PJSIP_SC_PROXY_AUTHENTICATION_REQUIRED) &&
!client_state->auth_rejection_permanent) ||
PJSIP_IS_STATUS_IN_CLASS(code, 600)) {
return 1;
} else {
return 0;
}
}
static void schedule_retry(struct registration_response *response, unsigned int interval,
const char *server_uri, const char *client_uri)
{
update_client_state_status(response->client_state, SIP_REGISTRATION_REJECTED_TEMPORARY);
schedule_registration(response->client_state, interval);
if (response->rdata) {
ast_log(LOG_WARNING, "Temporal response '%d' received from '%s' on "
"registration attempt to '%s', retrying in '%u'\n",
response->code, server_uri, client_uri, interval);
} else {
ast_log(LOG_WARNING, "No response received from '%s' on "
"registration attempt to '%s', retrying in '%u'\n",
server_uri, client_uri, interval);
}
}
static int reregister_immediately_cb(void *obj)
{
struct sip_outbound_registration_state *state = obj;
if (state->client_state->status != SIP_REGISTRATION_REGISTERED) {
ao2_ref(state, -1);
return 0;
}
if (DEBUG_ATLEAST(1)) {
pjsip_regc_info info;
pjsip_regc_get_info(state->client_state->client, &info);
ast_log(LOG_DEBUG,
"Outbound registration transport to server '%.*s' from client '%.*s' shutdown\n",
(int) info.server_uri.slen, info.server_uri.ptr,
(int) info.client_uri.slen, info.client_uri.ptr);
}
cancel_registration(state->client_state);
ao2_ref(state->client_state, +1);
handle_client_registration(state->client_state);
ao2_ref(state, -1);
return 0;
}
/*!
* \internal
* \brief The reliable transport we registered using has shutdown.
* \since 13.18.0
*
* \param obj What is needed to initiate a reregister attempt.
*
* \note Normally executed by the pjsip monitor thread.
*/
static void registration_transport_shutdown_cb(void *obj)
{
const char *registration_name = obj;
struct sip_outbound_registration_state *state;
state = get_state(registration_name);
if (!state) {
/* Registration no longer exists or shutting down. */
return;
}
if (ast_sip_push_task(state->client_state->serializer, reregister_immediately_cb, state)) {
ao2_ref(state, -1);
}
}
static int monitor_matcher(void *a, void *b)
{
char *ma = a;
char *mb = b;
return strcmp(ma, mb) == 0;
}
pjsip_transport_events: Fix possible use after free on transport It was possible for a module that registered for transport monitor events to pass in a pjsip_transport that had already been freed. This caused pjsip_transport_events to crash when looking up the monitor for the transport. The fix is a two pronged approach. 1. We now increment the reference count on pjsip_transports when we create monitors for them, then decrement the count when the transport is going to be destroyed. 2. There are now APIs to register and unregister monitor callbacks by "transport key" which is a string concatenation of the remote ip address and port. This way the module needing to monitor the transport doesn't have to hold on to the transport object itself to unregister. It just has to save the transport_key. * Added the pjsip_transport reference increment and decrement. * Changed the internal transport monitor container key from the transport->obj_name (which may not be unique anyway) to the transport_key. * Added a helper macro AST_SIP_MAKE_REMOTE_IPADDR_PORT_STR() that fills a buffer with the transport_key using a passed-in pjsip_transport. * Added the following functions: ast_sip_transport_monitor_register_key ast_sip_transport_monitor_register_replace_key ast_sip_transport_monitor_unregister_key and marked their non-key counterparts as deprecated. * Updated res_pjsip_pubsub and res_pjsip_outbound_register to use the new "key" monitor functions. NOTE: res_pjsip_registrar also uses the transport monitor functionality but doesn't have a persistent object other than contact to store a transport key. At this time, it continues to use the non-key monitor functions. ASTERISK-30244 Change-Id: I1a20baf2a8643c272dcf819871d6c395f148f00b (cherry picked from commit 7684c9e907fb85f5c58b025d9e385ad2600f12a2)
2022-10-10 14:35:54 +00:00
static void registration_transport_monitor_setup(const char *transport_key, const char *registration_name)
{
char *monitor;
monitor = ao2_alloc_options(strlen(registration_name) + 1, NULL,
AO2_ALLOC_OPT_LOCK_NOLOCK);
if (!monitor) {
return;
}
strcpy(monitor, registration_name);/* Safe */
/*
* We'll ignore if the transport has already been shutdown before we
* register the monitor. We might get into a message spamming infinite
* loop of registration, shutdown, reregistration...
*/
pjsip_transport_events: Fix possible use after free on transport It was possible for a module that registered for transport monitor events to pass in a pjsip_transport that had already been freed. This caused pjsip_transport_events to crash when looking up the monitor for the transport. The fix is a two pronged approach. 1. We now increment the reference count on pjsip_transports when we create monitors for them, then decrement the count when the transport is going to be destroyed. 2. There are now APIs to register and unregister monitor callbacks by "transport key" which is a string concatenation of the remote ip address and port. This way the module needing to monitor the transport doesn't have to hold on to the transport object itself to unregister. It just has to save the transport_key. * Added the pjsip_transport reference increment and decrement. * Changed the internal transport monitor container key from the transport->obj_name (which may not be unique anyway) to the transport_key. * Added a helper macro AST_SIP_MAKE_REMOTE_IPADDR_PORT_STR() that fills a buffer with the transport_key using a passed-in pjsip_transport. * Added the following functions: ast_sip_transport_monitor_register_key ast_sip_transport_monitor_register_replace_key ast_sip_transport_monitor_unregister_key and marked their non-key counterparts as deprecated. * Updated res_pjsip_pubsub and res_pjsip_outbound_register to use the new "key" monitor functions. NOTE: res_pjsip_registrar also uses the transport monitor functionality but doesn't have a persistent object other than contact to store a transport key. At this time, it continues to use the non-key monitor functions. ASTERISK-30244 Change-Id: I1a20baf2a8643c272dcf819871d6c395f148f00b (cherry picked from commit 7684c9e907fb85f5c58b025d9e385ad2600f12a2)
2022-10-10 14:35:54 +00:00
ast_sip_transport_monitor_register_replace_key(transport_key, registration_transport_shutdown_cb,
monitor, monitor_matcher);
ao2_ref(monitor, -1);
}
res_pjsip: Implement additional SIP RFCs for Google Voice trunk compatability This change implements a few different generic things which were brought on by Google Voice SIP. 1. The concept of flow transports have been introduced. These are configurable transports in pjsip.conf which can be used to reference a flow of signaling to a target. These have runtime configuration that can be changed by the signaling itself (such as Service-Routes and P-Preferred-Identity). When used these guarantee an individual connection (in the case of TCP or TLS) even if multiple flow transports exist to the same target. 2. Service-Routes (RFC 3608) support has been added to the outbound registration module which when received will be stored on the flow transport and used for requests referencing it. 3. P-Associated-URI / P-Preferred-Identity (RFC 3325) support has been added to the outbound registration module. If a P-Associated-URI header is received it will be used on requests as the P-Preferred-Identity. 4. Configurable outbound extension support has been added to the outbound registration module. When set the extension will be placed in the Supported header. 5. Header parameters can now be configured on an outbound registration which will be placed in the Contact header. 6. Google specific OAuth / Bearer token authentication (draft-ietf-sipcore-sip-authn-02) has been added to the outbound registration module. All functionality changes are controlled by pjsip.conf configuration options and do not affect non-configured pjsip endpoints otherwise. ASTERISK-27971 #close Change-Id: Id214c2d1c550a41fcf564b7df8f3da7be565bd58
2018-07-18 12:45:26 +00:00
static void save_response_fields_to_transport(struct registration_response *response)
{
static const pj_str_t associated_uri_str = { "P-Associated-URI", 16 };
static const pj_str_t service_route_str = { "Service-Route", 13 };
pjsip_hdr *header = NULL;
pjsip_msg *msg = response->rdata->msg_info.msg;
struct ast_sip_service_route_vector *service_routes = NULL;
/* If no transport is specified then we can't update any */
if (ast_strlen_zero(response->client_state->transport_name)) {
return;
}
ast_sip_transport_state_set_transport(response->client_state->transport_name, response->rdata->tp_info.transport);
while ((header = pjsip_msg_find_hdr_by_name(msg, &service_route_str, header ? header->next : NULL))) {
char *service_route;
size_t size;
/* The below code takes the approach that if we can't store all service routes then we
* store none at all. This gives a predictable failure condition instead of storing a
* partial list and having partial route headers.
*/
size = pj_strlen(&((pjsip_generic_string_hdr*)header)->hvalue) + 1;
service_route = ast_malloc(size);
if (!service_route) {
if (service_routes) {
ast_sip_service_route_vector_destroy(service_routes);
service_routes = NULL;
}
break;
}
ast_copy_pj_str(service_route, &((pjsip_generic_string_hdr*)header)->hvalue, size);
if (!service_routes) {
service_routes = ast_sip_service_route_vector_alloc();
if (!service_routes) {
ast_free(service_route);
break;
}
}
if (AST_VECTOR_APPEND(service_routes, service_route)) {
ast_free(service_route);
ast_sip_service_route_vector_destroy(service_routes);
service_routes = NULL;
break;
}
}
/* If any service routes were handled then store them on the transport */
if (service_routes) {
ast_sip_transport_state_set_service_routes(response->client_state->transport_name, service_routes);
}
/* If an associated URI is present in the response we need to use it on any outgoing
* traffic on the transport.
*/
header = pjsip_msg_find_hdr_by_name(msg, &associated_uri_str, NULL);
if (header) {
char value[pj_strlen(&((pjsip_generic_string_hdr*)header)->hvalue) + 1];
ast_copy_pj_str(value, &((pjsip_generic_string_hdr*)header)->hvalue, sizeof(value));
ast_sip_transport_state_set_preferred_identity(response->client_state->transport_name, value);
}
}
/*! \brief Callback function for handling a response to a registration attempt */
static int handle_registration_response(void *data)
{
res_pjsip_outbound_registration.c: Fix handle_client_state_destruction() refs * handle_client_state_destruction() must always be passed a ref to client_state because it will always unref client_state. handle_registration_response() was not passing a client_state ref. * Made the final un-REGISTER message get sent normally using the pjproject register control structure in handle_client_state_destruction(). The previous code attempted to short circuit the response handling for the module to unload. That doesn't work for a couple reasons. One, pjsip_regc_send() may call the registered callback before it returns and unbalance the client_state ref count. Two, the registered callback handles any authentication for the un-REGISTER message. * Made the distinction between internal registration state and external registration status with sip_outbound_registration_status_str(). This is necessary to avoid altering documented AMI messages with internal changes. * Removed references to client_state->client outside of the serializer thread. When handle_client_state_destruction() destroys the pjproject register control structure that memory is freed and cannot be referenced anymore. These accesses were to provide information for debug and off-nominal warning messages. * In sip_outbound_registration_timer_cb() you should not access entry->id after unrefing client_state because the passed in entry is normally pointing to the timer entry in the client_state object. ASTERISK-24907 Reported by: Kevin Harwell Change-Id: Ia7b446d8644b6b4550ef5bea49527671de65183f
2015-06-16 20:06:22 +00:00
struct registration_response *response = data;
pjsip_regc_info info;
char server_uri[PJSIP_MAX_URL_SIZE];
char client_uri[PJSIP_MAX_URL_SIZE];
if (response->client_state->status == SIP_REGISTRATION_STOPPED) {
res_pjsip_outbound_registration.c: Fix handle_client_state_destruction() refs * handle_client_state_destruction() must always be passed a ref to client_state because it will always unref client_state. handle_registration_response() was not passing a client_state ref. * Made the final un-REGISTER message get sent normally using the pjproject register control structure in handle_client_state_destruction(). The previous code attempted to short circuit the response handling for the module to unload. That doesn't work for a couple reasons. One, pjsip_regc_send() may call the registered callback before it returns and unbalance the client_state ref count. Two, the registered callback handles any authentication for the un-REGISTER message. * Made the distinction between internal registration state and external registration status with sip_outbound_registration_status_str(). This is necessary to avoid altering documented AMI messages with internal changes. * Removed references to client_state->client outside of the serializer thread. When handle_client_state_destruction() destroys the pjproject register control structure that memory is freed and cannot be referenced anymore. These accesses were to provide information for debug and off-nominal warning messages. * In sip_outbound_registration_timer_cb() you should not access entry->id after unrefing client_state because the passed in entry is normally pointing to the timer entry in the client_state object. ASTERISK-24907 Reported by: Kevin Harwell Change-Id: Ia7b446d8644b6b4550ef5bea49527671de65183f
2015-06-16 20:06:22 +00:00
ao2_ref(response, -1);
return 0;
}
pjsip_regc_get_info(response->client_state->client, &info);
ast_copy_pj_str(server_uri, &info.server_uri, sizeof(server_uri));
ast_copy_pj_str(client_uri, &info.client_uri, sizeof(client_uri));
response->client_state->last_status_code = response->code;
ast_debug(1, "Processing REGISTER response %d from server '%s' for client '%s'\n",
response->code, server_uri, client_uri);
if (response->code == 408 || response->code == 503) {
if ((ast_sip_failover_request(response->old_request))) {
int res = registration_client_send(response->client_state, response->old_request);
/* The tdata ref was stolen */
response->old_request = NULL;
if (res == PJ_SUCCESS) {
ao2_ref(response, -1);
return 0;
}
}
} else if ((response->code == 401 || response->code == 407 || response->code == 494)
&& (!response->client_state->auth_attempted
|| response->rdata->msg_info.cseq->cseq != response->client_state->auth_cseq)) {
int res;
pjsip_cseq_hdr *cseq_hdr;
pjsip_tx_data *tdata;
if (response->client_state->security_negotiation == AST_SIP_SECURITY_NEG_MEDIASEC) {
struct sip_outbound_registration *reg = NULL;
struct ast_sip_endpoint *endpt = NULL;
struct ao2_container *contact_container = NULL;
pjsip_generic_string_hdr *header;
struct pjsip_generic_string_hdr_vector header_vector;
static const pj_str_t security_server = { "Security-Server", 15 };
if ((reg = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "registration",
response->client_state->registration_name)) && reg->endpoint &&
(endpt = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "endpoint", reg->endpoint))) {
/* Retrieve all contacts associated with aors from this endpoint (if set). */
contact_container = ast_sip_location_retrieve_contacts_from_aor_list(endpt->aors);
}
/* Add server list of security mechanism to client_state and contact status if exists. */
AST_VECTOR_INIT(&header_vector, 1);
header = pjsip_msg_find_hdr_by_name(response->rdata->msg_info.msg, &security_server, NULL);
for (; header;
header = pjsip_msg_find_hdr_by_name(response->rdata->msg_info.msg, &security_server, header->next)) {
AST_VECTOR_APPEND(&header_vector, header);
ast_sip_header_to_security_mechanism(header, &response->client_state->server_security_mechanisms);
}
if (contact_container) {
/* Add server security mechanisms to contact status of all associated contacts to be able to send correct
* Security-Verify headers on subsequent non-REGISTER requests through this outbound registration.
*/
ao2_callback(contact_container, OBJ_NODATA, contact_add_security_headers_to_status, &header_vector);
ao2_cleanup(contact_container);
}
AST_VECTOR_FREE(&header_vector);
ao2_cleanup(endpt);
ao2_cleanup(reg);
}
if (response->code == 494) {
update_client_state_status(response->client_state, SIP_REGISTRATION_REJECTED_TEMPORARY);
response->client_state->retries++;
schedule_registration(response->client_state, 0);
ao2_ref(response, -1);
return 0;
} else if (!ast_sip_create_request_with_auth(&response->client_state->outbound_auths,
response->rdata, response->old_request, &tdata)) {
response->client_state->auth_attempted = 1;
ast_debug(1, "Sending authenticated REGISTER to server '%s' from client '%s'\n",
server_uri, client_uri);
pjsip_tx_data_add_ref(tdata);
res = registration_client_send(response->client_state, tdata);
/* Save the cseq that actually got sent. */
cseq_hdr = (pjsip_cseq_hdr *) pjsip_msg_find_hdr(tdata->msg, PJSIP_H_CSEQ,
NULL);
response->client_state->auth_cseq = cseq_hdr->cseq;
pjsip_tx_data_dec_ref(tdata);
if (res == PJ_SUCCESS) {
res_pjsip_outbound_registration.c: Fix handle_client_state_destruction() refs * handle_client_state_destruction() must always be passed a ref to client_state because it will always unref client_state. handle_registration_response() was not passing a client_state ref. * Made the final un-REGISTER message get sent normally using the pjproject register control structure in handle_client_state_destruction(). The previous code attempted to short circuit the response handling for the module to unload. That doesn't work for a couple reasons. One, pjsip_regc_send() may call the registered callback before it returns and unbalance the client_state ref count. Two, the registered callback handles any authentication for the un-REGISTER message. * Made the distinction between internal registration state and external registration status with sip_outbound_registration_status_str(). This is necessary to avoid altering documented AMI messages with internal changes. * Removed references to client_state->client outside of the serializer thread. When handle_client_state_destruction() destroys the pjproject register control structure that memory is freed and cannot be referenced anymore. These accesses were to provide information for debug and off-nominal warning messages. * In sip_outbound_registration_timer_cb() you should not access entry->id after unrefing client_state because the passed in entry is normally pointing to the timer entry in the client_state object. ASTERISK-24907 Reported by: Kevin Harwell Change-Id: Ia7b446d8644b6b4550ef5bea49527671de65183f
2015-06-16 20:06:22 +00:00
ao2_ref(response, -1);
return 0;
}
} else {
ast_log(LOG_WARNING, "Failed to create authenticated REGISTER request to server '%s' from client '%s'\n",
server_uri, client_uri);
}
/* Otherwise, fall through so the failure is processed appropriately */
}
response->client_state->auth_attempted = 0;
if (PJSIP_IS_STATUS_IN_CLASS(response->code, 200)) {
/* Check if this is in regards to registering or unregistering */
if (response->expiration) {
int next_registration_round;
/* If the registration went fine simply reschedule registration for the future */
ast_debug(1, "Outbound registration to '%s' with client '%s' successful\n", server_uri, client_uri);
update_client_state_status(response->client_state, SIP_REGISTRATION_REGISTERED);
response->client_state->retries = 0;
next_registration_round = response->expiration - REREGISTER_BUFFER_TIME;
if (next_registration_round < 0) {
/* Re-register immediately. */
next_registration_round = 0;
}
schedule_registration(response->client_state, next_registration_round);
/* See if we should monitor for transport shutdown */
pjsip_transport_events: Fix possible use after free on transport It was possible for a module that registered for transport monitor events to pass in a pjsip_transport that had already been freed. This caused pjsip_transport_events to crash when looking up the monitor for the transport. The fix is a two pronged approach. 1. We now increment the reference count on pjsip_transports when we create monitors for them, then decrement the count when the transport is going to be destroyed. 2. There are now APIs to register and unregister monitor callbacks by "transport key" which is a string concatenation of the remote ip address and port. This way the module needing to monitor the transport doesn't have to hold on to the transport object itself to unregister. It just has to save the transport_key. * Added the pjsip_transport reference increment and decrement. * Changed the internal transport monitor container key from the transport->obj_name (which may not be unique anyway) to the transport_key. * Added a helper macro AST_SIP_MAKE_REMOTE_IPADDR_PORT_STR() that fills a buffer with the transport_key using a passed-in pjsip_transport. * Added the following functions: ast_sip_transport_monitor_register_key ast_sip_transport_monitor_register_replace_key ast_sip_transport_monitor_unregister_key and marked their non-key counterparts as deprecated. * Updated res_pjsip_pubsub and res_pjsip_outbound_register to use the new "key" monitor functions. NOTE: res_pjsip_registrar also uses the transport monitor functionality but doesn't have a persistent object other than contact to store a transport key. At this time, it continues to use the non-key monitor functions. ASTERISK-30244 Change-Id: I1a20baf2a8643c272dcf819871d6c395f148f00b (cherry picked from commit 7684c9e907fb85f5c58b025d9e385ad2600f12a2)
2022-10-10 14:35:54 +00:00
if (PJSIP_TRANSPORT_IS_RELIABLE(response->rdata->tp_info.transport)) {
registration_transport_monitor_setup(response->transport_key,
response->client_state->registration_name);
}
} else {
ast_debug(1, "Outbound unregistration to '%s' with client '%s' successful\n", server_uri, client_uri);
update_client_state_status(response->client_state, SIP_REGISTRATION_UNREGISTERED);
pjsip_transport_events: Fix possible use after free on transport It was possible for a module that registered for transport monitor events to pass in a pjsip_transport that had already been freed. This caused pjsip_transport_events to crash when looking up the monitor for the transport. The fix is a two pronged approach. 1. We now increment the reference count on pjsip_transports when we create monitors for them, then decrement the count when the transport is going to be destroyed. 2. There are now APIs to register and unregister monitor callbacks by "transport key" which is a string concatenation of the remote ip address and port. This way the module needing to monitor the transport doesn't have to hold on to the transport object itself to unregister. It just has to save the transport_key. * Added the pjsip_transport reference increment and decrement. * Changed the internal transport monitor container key from the transport->obj_name (which may not be unique anyway) to the transport_key. * Added a helper macro AST_SIP_MAKE_REMOTE_IPADDR_PORT_STR() that fills a buffer with the transport_key using a passed-in pjsip_transport. * Added the following functions: ast_sip_transport_monitor_register_key ast_sip_transport_monitor_register_replace_key ast_sip_transport_monitor_unregister_key and marked their non-key counterparts as deprecated. * Updated res_pjsip_pubsub and res_pjsip_outbound_register to use the new "key" monitor functions. NOTE: res_pjsip_registrar also uses the transport monitor functionality but doesn't have a persistent object other than contact to store a transport key. At this time, it continues to use the non-key monitor functions. ASTERISK-30244 Change-Id: I1a20baf2a8643c272dcf819871d6c395f148f00b (cherry picked from commit 7684c9e907fb85f5c58b025d9e385ad2600f12a2)
2022-10-10 14:35:54 +00:00
if (PJSIP_TRANSPORT_IS_RELIABLE(response->rdata->tp_info.transport)) {
ast_sip_transport_monitor_unregister_key(response->transport_key,
registration_transport_shutdown_cb, response->client_state->registration_name,
monitor_matcher);
}
}
res_pjsip: Implement additional SIP RFCs for Google Voice trunk compatability This change implements a few different generic things which were brought on by Google Voice SIP. 1. The concept of flow transports have been introduced. These are configurable transports in pjsip.conf which can be used to reference a flow of signaling to a target. These have runtime configuration that can be changed by the signaling itself (such as Service-Routes and P-Preferred-Identity). When used these guarantee an individual connection (in the case of TCP or TLS) even if multiple flow transports exist to the same target. 2. Service-Routes (RFC 3608) support has been added to the outbound registration module which when received will be stored on the flow transport and used for requests referencing it. 3. P-Associated-URI / P-Preferred-Identity (RFC 3325) support has been added to the outbound registration module. If a P-Associated-URI header is received it will be used on requests as the P-Preferred-Identity. 4. Configurable outbound extension support has been added to the outbound registration module. When set the extension will be placed in the Supported header. 5. Header parameters can now be configured on an outbound registration which will be placed in the Contact header. 6. Google specific OAuth / Bearer token authentication (draft-ietf-sipcore-sip-authn-02) has been added to the outbound registration module. All functionality changes are controlled by pjsip.conf configuration options and do not affect non-configured pjsip endpoints otherwise. ASTERISK-27971 #close Change-Id: Id214c2d1c550a41fcf564b7df8f3da7be565bd58
2018-07-18 12:45:26 +00:00
save_response_fields_to_transport(response);
res_pjsip_outbound_registration.c: Fix handle_client_state_destruction() refs * handle_client_state_destruction() must always be passed a ref to client_state because it will always unref client_state. handle_registration_response() was not passing a client_state ref. * Made the final un-REGISTER message get sent normally using the pjproject register control structure in handle_client_state_destruction(). The previous code attempted to short circuit the response handling for the module to unload. That doesn't work for a couple reasons. One, pjsip_regc_send() may call the registered callback before it returns and unbalance the client_state ref count. Two, the registered callback handles any authentication for the un-REGISTER message. * Made the distinction between internal registration state and external registration status with sip_outbound_registration_status_str(). This is necessary to avoid altering documented AMI messages with internal changes. * Removed references to client_state->client outside of the serializer thread. When handle_client_state_destruction() destroys the pjproject register control structure that memory is freed and cannot be referenced anymore. These accesses were to provide information for debug and off-nominal warning messages. * In sip_outbound_registration_timer_cb() you should not access entry->id after unrefing client_state because the passed in entry is normally pointing to the timer entry in the client_state object. ASTERISK-24907 Reported by: Kevin Harwell Change-Id: Ia7b446d8644b6b4550ef5bea49527671de65183f
2015-06-16 20:06:22 +00:00
} else if (response->client_state->destroy) {
/* We need to deal with the pending destruction instead. */
} else if (response->retry_after) {
/* If we have been instructed to retry after a period of time, schedule it as such */
schedule_retry(response, response->retry_after, server_uri, client_uri);
} else if (response->client_state->retry_interval
&& sip_outbound_registration_is_temporal(response->code, response->client_state)) {
if (response->client_state->retries == response->client_state->max_retries) {
/* If we received enough temporal responses to exceed our maximum give up permanently */
update_client_state_status(response->client_state, SIP_REGISTRATION_REJECTED_PERMANENT);
ast_log(LOG_WARNING, "Maximum retries reached when attempting outbound registration to '%s' with client '%s', stopping registration attempt\n",
server_uri, client_uri);
} else {
/* On the other hand if we can still try some more do so */
response->client_state->retries++;
schedule_retry(response, response->client_state->retry_interval, server_uri, client_uri);
}
} else {
if (response->code == 403
&& response->client_state->forbidden_retry_interval
&& response->client_state->retries < response->client_state->max_retries) {
/* A forbidden response retry interval is configured and there are retries remaining */
update_client_state_status(response->client_state, SIP_REGISTRATION_REJECTED_TEMPORARY);
response->client_state->retries++;
schedule_registration(response->client_state, response->client_state->forbidden_retry_interval);
ast_log(LOG_WARNING, "403 Forbidden fatal response received from '%s' on registration attempt to '%s', retrying in '%u' seconds\n",
server_uri, client_uri, response->client_state->forbidden_retry_interval);
} else if (response->client_state->fatal_retry_interval
&& response->client_state->retries < response->client_state->max_retries) {
/* Some kind of fatal failure response received, so retry according to configured interval */
update_client_state_status(response->client_state, SIP_REGISTRATION_REJECTED_TEMPORARY);
response->client_state->retries++;
schedule_registration(response->client_state, response->client_state->fatal_retry_interval);
ast_log(LOG_WARNING, "'%d' fatal response received from '%s' on registration attempt to '%s', retrying in '%u' seconds\n",
response->code, server_uri, client_uri, response->client_state->fatal_retry_interval);
} else {
/* Finally if there's no hope of registering give up */
update_client_state_status(response->client_state, SIP_REGISTRATION_REJECTED_PERMANENT);
if (response->rdata) {
ast_log(LOG_WARNING, "Fatal response '%d' received from '%s' on registration attempt to '%s', stopping outbound registration\n",
response->code, server_uri, client_uri);
} else {
ast_log(LOG_WARNING, "Fatal registration attempt to '%s', stopping outbound registration\n", client_uri);
}
}
}
ast_system_publish_registry("PJSIP", client_uri, server_uri,
res_pjsip_outbound_registration.c: Fix handle_client_state_destruction() refs * handle_client_state_destruction() must always be passed a ref to client_state because it will always unref client_state. handle_registration_response() was not passing a client_state ref. * Made the final un-REGISTER message get sent normally using the pjproject register control structure in handle_client_state_destruction(). The previous code attempted to short circuit the response handling for the module to unload. That doesn't work for a couple reasons. One, pjsip_regc_send() may call the registered callback before it returns and unbalance the client_state ref count. Two, the registered callback handles any authentication for the un-REGISTER message. * Made the distinction between internal registration state and external registration status with sip_outbound_registration_status_str(). This is necessary to avoid altering documented AMI messages with internal changes. * Removed references to client_state->client outside of the serializer thread. When handle_client_state_destruction() destroys the pjproject register control structure that memory is freed and cannot be referenced anymore. These accesses were to provide information for debug and off-nominal warning messages. * In sip_outbound_registration_timer_cb() you should not access entry->id after unrefing client_state because the passed in entry is normally pointing to the timer entry in the client_state object. ASTERISK-24907 Reported by: Kevin Harwell Change-Id: Ia7b446d8644b6b4550ef5bea49527671de65183f
2015-06-16 20:06:22 +00:00
sip_outbound_registration_status_str(response->client_state->status), NULL);
if (response->client_state->destroy) {
res_pjsip_outbound_registration.c: Fix handle_client_state_destruction() refs * handle_client_state_destruction() must always be passed a ref to client_state because it will always unref client_state. handle_registration_response() was not passing a client_state ref. * Made the final un-REGISTER message get sent normally using the pjproject register control structure in handle_client_state_destruction(). The previous code attempted to short circuit the response handling for the module to unload. That doesn't work for a couple reasons. One, pjsip_regc_send() may call the registered callback before it returns and unbalance the client_state ref count. Two, the registered callback handles any authentication for the un-REGISTER message. * Made the distinction between internal registration state and external registration status with sip_outbound_registration_status_str(). This is necessary to avoid altering documented AMI messages with internal changes. * Removed references to client_state->client outside of the serializer thread. When handle_client_state_destruction() destroys the pjproject register control structure that memory is freed and cannot be referenced anymore. These accesses were to provide information for debug and off-nominal warning messages. * In sip_outbound_registration_timer_cb() you should not access entry->id after unrefing client_state because the passed in entry is normally pointing to the timer entry in the client_state object. ASTERISK-24907 Reported by: Kevin Harwell Change-Id: Ia7b446d8644b6b4550ef5bea49527671de65183f
2015-06-16 20:06:22 +00:00
/* We have a pending deferred destruction to complete now. */
ao2_ref(response->client_state, +1);
handle_client_state_destruction(response->client_state);
}
res_pjsip_outbound_registration.c: Fix handle_client_state_destruction() refs * handle_client_state_destruction() must always be passed a ref to client_state because it will always unref client_state. handle_registration_response() was not passing a client_state ref. * Made the final un-REGISTER message get sent normally using the pjproject register control structure in handle_client_state_destruction(). The previous code attempted to short circuit the response handling for the module to unload. That doesn't work for a couple reasons. One, pjsip_regc_send() may call the registered callback before it returns and unbalance the client_state ref count. Two, the registered callback handles any authentication for the un-REGISTER message. * Made the distinction between internal registration state and external registration status with sip_outbound_registration_status_str(). This is necessary to avoid altering documented AMI messages with internal changes. * Removed references to client_state->client outside of the serializer thread. When handle_client_state_destruction() destroys the pjproject register control structure that memory is freed and cannot be referenced anymore. These accesses were to provide information for debug and off-nominal warning messages. * In sip_outbound_registration_timer_cb() you should not access entry->id after unrefing client_state because the passed in entry is normally pointing to the timer entry in the client_state object. ASTERISK-24907 Reported by: Kevin Harwell Change-Id: Ia7b446d8644b6b4550ef5bea49527671de65183f
2015-06-16 20:06:22 +00:00
ao2_ref(response, -1);
return 0;
}
/*! \brief Callback function for outbound registration client */
static void sip_outbound_registration_response_cb(struct pjsip_regc_cbparam *param)
{
res_pjsip_outbound_registration.c: Fix handle_client_state_destruction() refs * handle_client_state_destruction() must always be passed a ref to client_state because it will always unref client_state. handle_registration_response() was not passing a client_state ref. * Made the final un-REGISTER message get sent normally using the pjproject register control structure in handle_client_state_destruction(). The previous code attempted to short circuit the response handling for the module to unload. That doesn't work for a couple reasons. One, pjsip_regc_send() may call the registered callback before it returns and unbalance the client_state ref count. Two, the registered callback handles any authentication for the un-REGISTER message. * Made the distinction between internal registration state and external registration status with sip_outbound_registration_status_str(). This is necessary to avoid altering documented AMI messages with internal changes. * Removed references to client_state->client outside of the serializer thread. When handle_client_state_destruction() destroys the pjproject register control structure that memory is freed and cannot be referenced anymore. These accesses were to provide information for debug and off-nominal warning messages. * In sip_outbound_registration_timer_cb() you should not access entry->id after unrefing client_state because the passed in entry is normally pointing to the timer entry in the client_state object. ASTERISK-24907 Reported by: Kevin Harwell Change-Id: Ia7b446d8644b6b4550ef5bea49527671de65183f
2015-06-16 20:06:22 +00:00
struct sip_outbound_registration_client_state *client_state = param->token;
struct registration_response *response;
int *callback_invoked;
callback_invoked = ast_threadstorage_get(&register_callback_invoked, sizeof(int));
ast_assert(callback_invoked != NULL);
ast_assert(client_state != NULL);
*callback_invoked = 1;
response = ao2_alloc(sizeof(*response), registration_response_destroy);
if (!response) {
res_pjsip_outbound_registration.c: Fix handle_client_state_destruction() refs * handle_client_state_destruction() must always be passed a ref to client_state because it will always unref client_state. handle_registration_response() was not passing a client_state ref. * Made the final un-REGISTER message get sent normally using the pjproject register control structure in handle_client_state_destruction(). The previous code attempted to short circuit the response handling for the module to unload. That doesn't work for a couple reasons. One, pjsip_regc_send() may call the registered callback before it returns and unbalance the client_state ref count. Two, the registered callback handles any authentication for the un-REGISTER message. * Made the distinction between internal registration state and external registration status with sip_outbound_registration_status_str(). This is necessary to avoid altering documented AMI messages with internal changes. * Removed references to client_state->client outside of the serializer thread. When handle_client_state_destruction() destroys the pjproject register control structure that memory is freed and cannot be referenced anymore. These accesses were to provide information for debug and off-nominal warning messages. * In sip_outbound_registration_timer_cb() you should not access entry->id after unrefing client_state because the passed in entry is normally pointing to the timer entry in the client_state object. ASTERISK-24907 Reported by: Kevin Harwell Change-Id: Ia7b446d8644b6b4550ef5bea49527671de65183f
2015-06-16 20:06:22 +00:00
ao2_ref(client_state, -1);
return;
}
response->code = param->code;
response->expiration = param->expiration;
/*
* Transfer client_state reference to response so the
* nominal path will not dec the client_state ref in this
* pjproject callback thread.
*/
response->client_state = client_state;
ast_debug(1, "Received REGISTER response %d(%.*s)\n",
param->code, (int) param->reason.slen, param->reason.ptr);
if (param->rdata) {
struct pjsip_retry_after_hdr *retry_after;
pjsip_transaction *tsx;
retry_after = pjsip_msg_find_hdr(param->rdata->msg_info.msg, PJSIP_H_RETRY_AFTER,
NULL);
response->retry_after = retry_after ? retry_after->ivalue : 0;
res_pjsip_outbound_registration: Fix SRV failover on timeout In order to retry outbound registrations for some situations, we need access to the tdata from the original request. For instance, for 401/407 responses we need it to properly construct the subsequent request with the authentication. We also need it if we're iterating over a DNS SRV response record set so we can skip entries we've already tried. We've been getting the tdata from the server response rdata and transaction but that only works for the failures where there was actually a response (4XX, 5XX, etc). For timeouts there's no response and therefore no rdata or transaction from which to get the tdata. When processing a single A/AAAA record for a server, this wasn't an issue as we just retried that same server after the retry timer expired. If we got an SRV record set for the server though, without the state from the tdata, we just kept trying the first entry in the set repeatedly instead of skipping to the next one in the list. * Added a "last_tdata" member to the client state structure to keep track of the sent tdata. * Updated registration_client_send() to save the tdata it used into the client_state. * Updated sip_outbound_registration_response_cb() to use the tdata saved in client_state when we don't get a response from the server. We still use the tdata from the transaction when we DO get a response from the server so we can properly handle 4XX responses where our new request depends on it. General note on timeouts: Although res_pjsip_outbound_registration skips to the next record immediately when a timeout occurs during SRV set traversal, it's pjproject that determines how long to wait before a timeout is declared. As with other SIP message types, pjproject will continue trying the same server at an interval specified by "timer_t1" until "timer_b" expires. Both of those timers are set in the pjsip.conf "system" section. ASTERISK-28746 Change-Id: I199b8274392d17661dd3ce3b4d69a3968368fa06
2020-02-13 19:39:58 +00:00
/*
* If we got a response from the server, we have to use the tdata
* from the transaction, not the tdata saved when we sent the
* request. If we use the saved tdata, we won't process responses
* like 423 Interval Too Brief correctly and we'll wind up sending
* the bad Expires value again.
*/
pjsip_tx_data_dec_ref(client_state->last_tdata);
tsx = pjsip_rdata_get_tsx(param->rdata);
response->old_request = tsx->last_tx;
pjsip_tx_data_add_ref(response->old_request);
pjsip_rx_data_clone(param->rdata, 0, &response->rdata);
pjsip_transport_events: Fix possible use after free on transport It was possible for a module that registered for transport monitor events to pass in a pjsip_transport that had already been freed. This caused pjsip_transport_events to crash when looking up the monitor for the transport. The fix is a two pronged approach. 1. We now increment the reference count on pjsip_transports when we create monitors for them, then decrement the count when the transport is going to be destroyed. 2. There are now APIs to register and unregister monitor callbacks by "transport key" which is a string concatenation of the remote ip address and port. This way the module needing to monitor the transport doesn't have to hold on to the transport object itself to unregister. It just has to save the transport_key. * Added the pjsip_transport reference increment and decrement. * Changed the internal transport monitor container key from the transport->obj_name (which may not be unique anyway) to the transport_key. * Added a helper macro AST_SIP_MAKE_REMOTE_IPADDR_PORT_STR() that fills a buffer with the transport_key using a passed-in pjsip_transport. * Added the following functions: ast_sip_transport_monitor_register_key ast_sip_transport_monitor_register_replace_key ast_sip_transport_monitor_unregister_key and marked their non-key counterparts as deprecated. * Updated res_pjsip_pubsub and res_pjsip_outbound_register to use the new "key" monitor functions. NOTE: res_pjsip_registrar also uses the transport monitor functionality but doesn't have a persistent object other than contact to store a transport key. At this time, it continues to use the non-key monitor functions. ASTERISK-30244 Change-Id: I1a20baf2a8643c272dcf819871d6c395f148f00b (cherry picked from commit 7684c9e907fb85f5c58b025d9e385ad2600f12a2)
2022-10-10 14:35:54 +00:00
AST_SIP_MAKE_REMOTE_IPADDR_PORT_STR(param->rdata->tp_info.transport,
response->transport_key);
res_pjsip_outbound_registration: Fix SRV failover on timeout In order to retry outbound registrations for some situations, we need access to the tdata from the original request. For instance, for 401/407 responses we need it to properly construct the subsequent request with the authentication. We also need it if we're iterating over a DNS SRV response record set so we can skip entries we've already tried. We've been getting the tdata from the server response rdata and transaction but that only works for the failures where there was actually a response (4XX, 5XX, etc). For timeouts there's no response and therefore no rdata or transaction from which to get the tdata. When processing a single A/AAAA record for a server, this wasn't an issue as we just retried that same server after the retry timer expired. If we got an SRV record set for the server though, without the state from the tdata, we just kept trying the first entry in the set repeatedly instead of skipping to the next one in the list. * Added a "last_tdata" member to the client state structure to keep track of the sent tdata. * Updated registration_client_send() to save the tdata it used into the client_state. * Updated sip_outbound_registration_response_cb() to use the tdata saved in client_state when we don't get a response from the server. We still use the tdata from the transaction when we DO get a response from the server so we can properly handle 4XX responses where our new request depends on it. General note on timeouts: Although res_pjsip_outbound_registration skips to the next record immediately when a timeout occurs during SRV set traversal, it's pjproject that determines how long to wait before a timeout is declared. As with other SIP message types, pjproject will continue trying the same server at an interval specified by "timer_t1" until "timer_b" expires. Both of those timers are set in the pjsip.conf "system" section. ASTERISK-28746 Change-Id: I199b8274392d17661dd3ce3b4d69a3968368fa06
2020-02-13 19:39:58 +00:00
} else {
/* old_request steals the reference */
response->old_request = client_state->last_tdata;
}
res_pjsip_outbound_registration: Fix SRV failover on timeout In order to retry outbound registrations for some situations, we need access to the tdata from the original request. For instance, for 401/407 responses we need it to properly construct the subsequent request with the authentication. We also need it if we're iterating over a DNS SRV response record set so we can skip entries we've already tried. We've been getting the tdata from the server response rdata and transaction but that only works for the failures where there was actually a response (4XX, 5XX, etc). For timeouts there's no response and therefore no rdata or transaction from which to get the tdata. When processing a single A/AAAA record for a server, this wasn't an issue as we just retried that same server after the retry timer expired. If we got an SRV record set for the server though, without the state from the tdata, we just kept trying the first entry in the set repeatedly instead of skipping to the next one in the list. * Added a "last_tdata" member to the client state structure to keep track of the sent tdata. * Updated registration_client_send() to save the tdata it used into the client_state. * Updated sip_outbound_registration_response_cb() to use the tdata saved in client_state when we don't get a response from the server. We still use the tdata from the transaction when we DO get a response from the server so we can properly handle 4XX responses where our new request depends on it. General note on timeouts: Although res_pjsip_outbound_registration skips to the next record immediately when a timeout occurs during SRV set traversal, it's pjproject that determines how long to wait before a timeout is declared. As with other SIP message types, pjproject will continue trying the same server at an interval specified by "timer_t1" until "timer_b" expires. Both of those timers are set in the pjsip.conf "system" section. ASTERISK-28746 Change-Id: I199b8274392d17661dd3ce3b4d69a3968368fa06
2020-02-13 19:39:58 +00:00
client_state->last_tdata = NULL;
/*
* Transfer response reference to serializer task so the
* nominal path will not dec the response ref in this
* pjproject callback thread.
*/
if (ast_sip_push_task(client_state->serializer, handle_registration_response, response)) {
ast_log(LOG_WARNING, "Failed to pass incoming registration response to threadpool\n");
ao2_cleanup(response);
}
}
/*! \brief Destructor function for registration state */
static void sip_outbound_registration_state_destroy(void *obj)
{
struct sip_outbound_registration_state *state = obj;
ast_debug(3, "Destroying registration state for registration to server '%s' from client '%s'\n",
state->registration ? state->registration->server_uri : "",
state->registration ? state->registration->client_uri : "");
ao2_cleanup(state->registration);
if (!state->client_state) {
/* Nothing to do */
} else if (!state->client_state->serializer) {
ao2_ref(state->client_state, -1);
} else if (ast_sip_push_task(state->client_state->serializer,
handle_client_state_destruction, state->client_state)) {
ast_log(LOG_WARNING, "Failed to pass outbound registration client destruction to threadpool\n");
ao2_ref(state->client_state, -1);
}
}
/*! \brief Destructor function for client registration state */
static void sip_outbound_registration_client_state_destroy(void *obj)
{
struct sip_outbound_registration_client_state *client_state = obj;
ast_statsd_log_string("PJSIP.registrations.count", AST_STATSD_GAUGE, "-1", 1.0);
ast_statsd_log_string_va("PJSIP.registrations.state.%s", AST_STATSD_GAUGE, "-1", 1.0,
sip_outbound_registration_status_str(client_state->status));
ast_taskprocessor_unreference(client_state->serializer);
ast_free(client_state->transport_name);
ast_free(client_state->registration_name);
ast_free(client_state->user_agent);
res_pjsip_outbound_registration: Fix SRV failover on timeout In order to retry outbound registrations for some situations, we need access to the tdata from the original request. For instance, for 401/407 responses we need it to properly construct the subsequent request with the authentication. We also need it if we're iterating over a DNS SRV response record set so we can skip entries we've already tried. We've been getting the tdata from the server response rdata and transaction but that only works for the failures where there was actually a response (4XX, 5XX, etc). For timeouts there's no response and therefore no rdata or transaction from which to get the tdata. When processing a single A/AAAA record for a server, this wasn't an issue as we just retried that same server after the retry timer expired. If we got an SRV record set for the server though, without the state from the tdata, we just kept trying the first entry in the set repeatedly instead of skipping to the next one in the list. * Added a "last_tdata" member to the client state structure to keep track of the sent tdata. * Updated registration_client_send() to save the tdata it used into the client_state. * Updated sip_outbound_registration_response_cb() to use the tdata saved in client_state when we don't get a response from the server. We still use the tdata from the transaction when we DO get a response from the server so we can properly handle 4XX responses where our new request depends on it. General note on timeouts: Although res_pjsip_outbound_registration skips to the next record immediately when a timeout occurs during SRV set traversal, it's pjproject that determines how long to wait before a timeout is declared. As with other SIP message types, pjproject will continue trying the same server at an interval specified by "timer_t1" until "timer_b" expires. Both of those timers are set in the pjsip.conf "system" section. ASTERISK-28746 Change-Id: I199b8274392d17661dd3ce3b4d69a3968368fa06
2020-02-13 19:39:58 +00:00
if (client_state->last_tdata) {
pjsip_tx_data_dec_ref(client_state->last_tdata);
}
}
/*! \brief Allocator function for registration state */
static struct sip_outbound_registration_state *sip_outbound_registration_state_alloc(struct sip_outbound_registration *registration)
{
struct sip_outbound_registration_state *state;
char tps_name[AST_TASKPROCESSOR_MAX_NAME + 1];
state = ao2_alloc(sizeof(*state), sip_outbound_registration_state_destroy);
if (!state) {
return NULL;
}
state->client_state = ao2_alloc(sizeof(*state->client_state),
sip_outbound_registration_client_state_destroy);
if (!state->client_state) {
ao2_cleanup(state);
return NULL;
}
state->client_state->status = SIP_REGISTRATION_UNREGISTERED;
pj_timer_entry_init(&state->client_state->timer, 0, state->client_state,
sip_outbound_registration_timer_cb);
state->client_state->transport_name = ast_strdup(registration->transport);
state->client_state->registration_name =
ast_strdup(ast_sorcery_object_get_id(registration));
state->client_state->user_agent = ast_strdup(registration->user_agent);
ast_statsd_log_string("PJSIP.registrations.count", AST_STATSD_GAUGE, "+1", 1.0);
ast_statsd_log_string_va("PJSIP.registrations.state.%s", AST_STATSD_GAUGE, "+1", 1.0,
sip_outbound_registration_status_str(state->client_state->status));
if (!state->client_state->transport_name
|| !state->client_state->registration_name) {
ao2_cleanup(state);
return NULL;
}
/* Create name with seq number appended. */
ast_taskprocessor_build_name(tps_name, sizeof(tps_name), "pjsip/outreg/%s",
ast_sorcery_object_get_id(registration));
state->client_state->serializer = ast_sip_create_serializer_group(tps_name,
shutdown_group);
if (!state->client_state->serializer) {
ao2_cleanup(state);
return NULL;
}
state->registration = ao2_bump(registration);
return state;
}
/*! \brief Destructor function for registration information */
static void sip_outbound_registration_destroy(void *obj)
{
struct sip_outbound_registration *registration = obj;
ast_sip_auth_vector_destroy(&registration->outbound_auths);
ast_sip_security_mechanisms_vector_destroy(&registration->security_mechanisms);
ast_string_field_free_memory(registration);
}
/*! \brief Allocator function for registration information */
static void *sip_outbound_registration_alloc(const char *name)
{
struct sip_outbound_registration *registration;
registration = ast_sorcery_generic_alloc(sizeof(*registration),
sip_outbound_registration_destroy);
if (!registration || ast_string_field_init(registration, 256)) {
ao2_cleanup(registration);
return NULL;
}
return registration;
}
/*! \brief Helper function which populates a pj_str_t with a contact header */
static int sip_dialog_create_contact(pj_pool_t *pool, pj_str_t *contact, const char *user,
res_pjsip: Implement additional SIP RFCs for Google Voice trunk compatability This change implements a few different generic things which were brought on by Google Voice SIP. 1. The concept of flow transports have been introduced. These are configurable transports in pjsip.conf which can be used to reference a flow of signaling to a target. These have runtime configuration that can be changed by the signaling itself (such as Service-Routes and P-Preferred-Identity). When used these guarantee an individual connection (in the case of TCP or TLS) even if multiple flow transports exist to the same target. 2. Service-Routes (RFC 3608) support has been added to the outbound registration module which when received will be stored on the flow transport and used for requests referencing it. 3. P-Associated-URI / P-Preferred-Identity (RFC 3325) support has been added to the outbound registration module. If a P-Associated-URI header is received it will be used on requests as the P-Preferred-Identity. 4. Configurable outbound extension support has been added to the outbound registration module. When set the extension will be placed in the Supported header. 5. Header parameters can now be configured on an outbound registration which will be placed in the Contact header. 6. Google specific OAuth / Bearer token authentication (draft-ietf-sipcore-sip-authn-02) has been added to the outbound registration module. All functionality changes are controlled by pjsip.conf configuration options and do not affect non-configured pjsip endpoints otherwise. ASTERISK-27971 #close Change-Id: Id214c2d1c550a41fcf564b7df8f3da7be565bd58
2018-07-18 12:45:26 +00:00
const pj_str_t *target, pjsip_tpselector *selector, const char *line, const char *header_params)
{
pj_str_t tmp, local_addr;
pjsip_uri *uri;
pjsip_sip_uri *sip_uri;
pjsip_transport_type_e type;
int local_port;
pj_strdup_with_null(pool, &tmp, target);
if (!(uri = pjsip_parse_uri(pool, tmp.ptr, tmp.slen, 0)) ||
(!PJSIP_URI_SCHEME_IS_SIP(uri) && !PJSIP_URI_SCHEME_IS_SIPS(uri))) {
return -1;
}
sip_uri = pjsip_uri_get_uri(uri);
type = pjsip_transport_get_type_from_name(&sip_uri->transport_param);
if (PJSIP_URI_SCHEME_IS_SIPS(sip_uri)) {
if (type == PJSIP_TRANSPORT_UNSPECIFIED
|| !(pjsip_transport_get_flag_from_type(type) & PJSIP_TRANSPORT_SECURE)) {
type = PJSIP_TRANSPORT_TLS;
}
} else if (!sip_uri->transport_param.slen) {
type = PJSIP_TRANSPORT_UDP;
} else if (type == PJSIP_TRANSPORT_UNSPECIFIED) {
return -1;
}
if (pj_strchr(&sip_uri->host, ':')) {
type |= PJSIP_TRANSPORT_IPV6;
}
if (pjsip_tpmgr_find_local_addr(pjsip_endpt_get_tpmgr(ast_sip_get_pjsip_endpoint()),
pool, type, selector, &local_addr, &local_port) != PJ_SUCCESS) {
return -1;
}
if (!pj_strchr(&sip_uri->host, ':') && pj_strchr(&local_addr, ':')) {
type |= PJSIP_TRANSPORT_IPV6;
}
contact->ptr = pj_pool_alloc(pool, PJSIP_MAX_URL_SIZE);
contact->slen = pj_ansi_snprintf(contact->ptr, PJSIP_MAX_URL_SIZE,
res_pjsip: Implement additional SIP RFCs for Google Voice trunk compatability This change implements a few different generic things which were brought on by Google Voice SIP. 1. The concept of flow transports have been introduced. These are configurable transports in pjsip.conf which can be used to reference a flow of signaling to a target. These have runtime configuration that can be changed by the signaling itself (such as Service-Routes and P-Preferred-Identity). When used these guarantee an individual connection (in the case of TCP or TLS) even if multiple flow transports exist to the same target. 2. Service-Routes (RFC 3608) support has been added to the outbound registration module which when received will be stored on the flow transport and used for requests referencing it. 3. P-Associated-URI / P-Preferred-Identity (RFC 3325) support has been added to the outbound registration module. If a P-Associated-URI header is received it will be used on requests as the P-Preferred-Identity. 4. Configurable outbound extension support has been added to the outbound registration module. When set the extension will be placed in the Supported header. 5. Header parameters can now be configured on an outbound registration which will be placed in the Contact header. 6. Google specific OAuth / Bearer token authentication (draft-ietf-sipcore-sip-authn-02) has been added to the outbound registration module. All functionality changes are controlled by pjsip.conf configuration options and do not affect non-configured pjsip endpoints otherwise. ASTERISK-27971 #close Change-Id: Id214c2d1c550a41fcf564b7df8f3da7be565bd58
2018-07-18 12:45:26 +00:00
"<%s:%s@%s%.*s%s:%d%s%s%s%s>%s%s",
((pjsip_transport_get_flag_from_type(type) & PJSIP_TRANSPORT_SECURE) && PJSIP_URI_SCHEME_IS_SIPS(uri)) ? "sips" : "sip",
user,
(type & PJSIP_TRANSPORT_IPV6) ? "[" : "",
(int)local_addr.slen,
local_addr.ptr,
(type & PJSIP_TRANSPORT_IPV6) ? "]" : "",
local_port,
(type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? ";transport=" : "",
(type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? pjsip_transport_get_type_name(type) : "",
!ast_strlen_zero(line) ? ";line=" : "",
res_pjsip: Implement additional SIP RFCs for Google Voice trunk compatability This change implements a few different generic things which were brought on by Google Voice SIP. 1. The concept of flow transports have been introduced. These are configurable transports in pjsip.conf which can be used to reference a flow of signaling to a target. These have runtime configuration that can be changed by the signaling itself (such as Service-Routes and P-Preferred-Identity). When used these guarantee an individual connection (in the case of TCP or TLS) even if multiple flow transports exist to the same target. 2. Service-Routes (RFC 3608) support has been added to the outbound registration module which when received will be stored on the flow transport and used for requests referencing it. 3. P-Associated-URI / P-Preferred-Identity (RFC 3325) support has been added to the outbound registration module. If a P-Associated-URI header is received it will be used on requests as the P-Preferred-Identity. 4. Configurable outbound extension support has been added to the outbound registration module. When set the extension will be placed in the Supported header. 5. Header parameters can now be configured on an outbound registration which will be placed in the Contact header. 6. Google specific OAuth / Bearer token authentication (draft-ietf-sipcore-sip-authn-02) has been added to the outbound registration module. All functionality changes are controlled by pjsip.conf configuration options and do not affect non-configured pjsip endpoints otherwise. ASTERISK-27971 #close Change-Id: Id214c2d1c550a41fcf564b7df8f3da7be565bd58
2018-07-18 12:45:26 +00:00
S_OR(line, ""),
!ast_strlen_zero(header_params) ? ";" : "",
S_OR(header_params, ""));
return 0;
}
/*!
* \internal
* \brief Check if a registration can be reused
*
* This checks if the existing outbound registration's configuration differs from a newly-applied
* outbound registration to see if the applied one.
*
* \param existing The pre-existing outbound registration
* \param applied The newly-created registration
*/
static int can_reuse_registration(struct sip_outbound_registration *existing,
struct sip_outbound_registration *applied)
{
int rc = 1;
struct ast_sorcery *sorcery = ast_sip_get_sorcery();
struct ast_variable *ve = ast_sorcery_objectset_create(sorcery, existing);
struct ast_variable *va = ast_sorcery_objectset_create(sorcery, applied);
struct ast_variable *vc = NULL;
if (ast_sorcery_changeset_create(ve, va, &vc) || vc != NULL) {
rc = 0;
ast_debug(4, "Registration '%s' changed. Can't re-use.\n", ast_sorcery_object_get_id(existing));
} else {
ast_debug(4, "Registration '%s' didn't change. Can re-use\n", ast_sorcery_object_get_id(existing));
}
ast_variables_destroy(ve);
ast_variables_destroy(va);
ast_variables_destroy(vc);
return rc;
}
/*! \brief Get google oauth2 access token using refresh token */
res_pjsip: Implement additional SIP RFCs for Google Voice trunk compatability This change implements a few different generic things which were brought on by Google Voice SIP. 1. The concept of flow transports have been introduced. These are configurable transports in pjsip.conf which can be used to reference a flow of signaling to a target. These have runtime configuration that can be changed by the signaling itself (such as Service-Routes and P-Preferred-Identity). When used these guarantee an individual connection (in the case of TCP or TLS) even if multiple flow transports exist to the same target. 2. Service-Routes (RFC 3608) support has been added to the outbound registration module which when received will be stored on the flow transport and used for requests referencing it. 3. P-Associated-URI / P-Preferred-Identity (RFC 3325) support has been added to the outbound registration module. If a P-Associated-URI header is received it will be used on requests as the P-Preferred-Identity. 4. Configurable outbound extension support has been added to the outbound registration module. When set the extension will be placed in the Supported header. 5. Header parameters can now be configured on an outbound registration which will be placed in the Contact header. 6. Google specific OAuth / Bearer token authentication (draft-ietf-sipcore-sip-authn-02) has been added to the outbound registration module. All functionality changes are controlled by pjsip.conf configuration options and do not affect non-configured pjsip endpoints otherwise. ASTERISK-27971 #close Change-Id: Id214c2d1c550a41fcf564b7df8f3da7be565bd58
2018-07-18 12:45:26 +00:00
static const char *fetch_google_access_token(const struct ast_sip_auth *auth)
{
char *cmd = NULL;
const char *token;
const char *url = "https://www.googleapis.com/oauth2/v3/token";
char buf[4096];
int res;
struct ast_json_error error;
struct ast_json *json;
/* set timeout to be shorter than default 180s (also checks func_curl is available) */
if (ast_func_write(NULL, "CURLOPT(conntimeout)", "10")) {
ast_log(LOG_ERROR, "CURL is unavailable. This is required for Google OAuth 2.0 authentication. Please ensure it is loaded.\n");
return NULL;
}
res = ast_asprintf(&cmd,
"CURL(%s,client_id=%s&client_secret=%s&refresh_token=%s&grant_type=refresh_token)",
url, auth->oauth_clientid, auth->oauth_secret, auth->refresh_token);
if (res < 0) {
return NULL;
}
ast_debug(2, "Performing Google OAuth 2.0 authentication using command: %s\n", cmd);
buf[0] = '\0';
res = ast_func_read(NULL, cmd, buf, sizeof(buf));
ast_free(cmd);
if (res) {
ast_log(LOG_ERROR, "Could not retrieve Google OAuth 2.0 authentication\n");
return NULL;
}
ast_debug(2, "Google OAuth 2.0 authentication returned: %s\n", buf);
json = ast_json_load_string(buf, &error);
if (!json) {
ast_log(LOG_ERROR, "Could not parse Google OAuth 2.0 authentication: %d(%d) %s: '%s'\n",
error.line, error.column, error.text, buf);
return NULL;
}
token = ast_json_string_get(ast_json_object_get(json, "access_token"));
if (!token) {
ast_log(LOG_ERROR, "Could not find Google OAuth 2.0 access_token in: '%s'\n",
buf);
}
token = ast_strdup(token);
ast_json_unref(json);
return token;
}
/*!
* \internal
* \brief Set pjsip registration context with any authentication credentials that need to be
* sent in the initial registration request
*
* \param regc The pjsip registration context
* \param auth_vector The vector of configured authentication credentials
*/
static int set_outbound_initial_authentication_credentials(pjsip_regc *regc,
const struct ast_sip_auth_vector *auth_vector)
{
size_t auth_size = AST_VECTOR_SIZE(auth_vector);
struct ast_sip_auth *auths[auth_size];
const char *access_token;
pjsip_cred_info auth_creds[1];
pjsip_auth_clt_pref prefs;
int res = 0;
int idx;
memset(auths, 0, sizeof(auths));
if (ast_sip_retrieve_auths(auth_vector, auths)) {
res = -1;
goto cleanup;
}
for (idx = 0; idx < auth_size; ++idx) {
switch (auths[idx]->type) {
case AST_SIP_AUTH_TYPE_GOOGLE_OAUTH:
pj_cstr(&auth_creds[0].username, auths[idx]->auth_user);
pj_cstr(&auth_creds[0].scheme, "Bearer");
pj_cstr(&auth_creds[0].realm, auths[idx]->realm);
ast_debug(2, "Obtaining Google OAuth access token\n");
access_token = fetch_google_access_token(auths[idx]);
if (!access_token) {
ast_log(LOG_WARNING, "Obtaining Google OAuth access token failed\n");
access_token = auths[idx]->auth_pass;
res = -1;
}
ast_debug(2, "Setting data to '%s'\n", access_token);
pj_cstr(&auth_creds[0].data, access_token);
auth_creds[0].data_type = PJSIP_CRED_DATA_PLAIN_PASSWD;
pjsip_regc_set_credentials(regc, 1, auth_creds);
/* for oauth, send auth without waiting for unauthorized response */
prefs.initial_auth = PJ_TRUE;
pj_cstr(&prefs.algorithm, "oauth");
pjsip_regc_set_prefs(regc, &prefs);
if (access_token != auths[idx]->auth_pass) {
ast_free((char *) access_token);
}
break;
default:
/* other cases handled after receiving auth rejection */
break;
}
}
cleanup:
ast_sip_cleanup_auths(auths, auth_size);
return res;
}
/*! \brief Helper function that allocates a pjsip registration client and configures it */
static int sip_outbound_registration_regc_alloc(void *data)
{
struct sip_outbound_registration_state *state = data;
RAII_VAR(struct sip_outbound_registration *, registration,
ao2_bump(state->registration), ao2_cleanup);
pj_pool_t *pool;
pj_str_t tmp;
pjsip_uri *uri;
pj_str_t server_uri, client_uri, contact_uri;
pjsip_tpselector selector = { .type = PJSIP_TPSELECTOR_NONE, };
pool = pjsip_endpt_create_pool(ast_sip_get_pjsip_endpoint(), "URI Validation", 256, 256);
if (!pool) {
ast_log(LOG_ERROR, "Could not create pool for URI validation on outbound registration '%s'\n",
ast_sorcery_object_get_id(registration));
return -1;
}
pj_strdup2_with_null(pool, &tmp, registration->server_uri);
uri = pjsip_parse_uri(pool, tmp.ptr, tmp.slen, 0);
if (!uri) {
ast_log(LOG_ERROR, "Invalid server URI '%s' specified on outbound registration '%s'\n",
registration->server_uri, ast_sorcery_object_get_id(registration));
pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
return -1;
}
pj_strdup2_with_null(pool, &tmp, registration->client_uri);
uri = pjsip_parse_uri(pool, tmp.ptr, tmp.slen, 0);
if (!uri) {
ast_log(LOG_ERROR, "Invalid client URI '%s' specified on outbound registration '%s'\n",
registration->client_uri, ast_sorcery_object_get_id(registration));
pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
return -1;
}
pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
res_pjsip_outbound_registration.c: Fix handle_client_state_destruction() refs * handle_client_state_destruction() must always be passed a ref to client_state because it will always unref client_state. handle_registration_response() was not passing a client_state ref. * Made the final un-REGISTER message get sent normally using the pjproject register control structure in handle_client_state_destruction(). The previous code attempted to short circuit the response handling for the module to unload. That doesn't work for a couple reasons. One, pjsip_regc_send() may call the registered callback before it returns and unbalance the client_state ref count. Two, the registered callback handles any authentication for the un-REGISTER message. * Made the distinction between internal registration state and external registration status with sip_outbound_registration_status_str(). This is necessary to avoid altering documented AMI messages with internal changes. * Removed references to client_state->client outside of the serializer thread. When handle_client_state_destruction() destroys the pjproject register control structure that memory is freed and cannot be referenced anymore. These accesses were to provide information for debug and off-nominal warning messages. * In sip_outbound_registration_timer_cb() you should not access entry->id after unrefing client_state because the passed in entry is normally pointing to the timer entry in the client_state object. ASTERISK-24907 Reported by: Kevin Harwell Change-Id: Ia7b446d8644b6b4550ef5bea49527671de65183f
2015-06-16 20:06:22 +00:00
ast_assert(state->client_state->client == NULL);
if (pjsip_regc_create(ast_sip_get_pjsip_endpoint(), state->client_state,
sip_outbound_registration_response_cb,
&state->client_state->client) != PJ_SUCCESS) {
return -1;
}
ast_sip_set_tpselector_from_transport_name(registration->transport, &selector);
pjsip_regc_set_transport(state->client_state->client, &selector);
if (!ast_strlen_zero(registration->outbound_proxy)) {
pjsip_route_hdr route_set, *route;
static const pj_str_t ROUTE_HNAME = { "Route", 5 };
pj_str_t tmp;
pj_list_init(&route_set);
pj_strdup2_with_null(pjsip_regc_get_pool(state->client_state->client), &tmp,
registration->outbound_proxy);
route = pjsip_parse_hdr(pjsip_regc_get_pool(state->client_state->client),
&ROUTE_HNAME, tmp.ptr, tmp.slen, NULL);
if (!route) {
res_pjsip: Implement additional SIP RFCs for Google Voice trunk compatability This change implements a few different generic things which were brought on by Google Voice SIP. 1. The concept of flow transports have been introduced. These are configurable transports in pjsip.conf which can be used to reference a flow of signaling to a target. These have runtime configuration that can be changed by the signaling itself (such as Service-Routes and P-Preferred-Identity). When used these guarantee an individual connection (in the case of TCP or TLS) even if multiple flow transports exist to the same target. 2. Service-Routes (RFC 3608) support has been added to the outbound registration module which when received will be stored on the flow transport and used for requests referencing it. 3. P-Associated-URI / P-Preferred-Identity (RFC 3325) support has been added to the outbound registration module. If a P-Associated-URI header is received it will be used on requests as the P-Preferred-Identity. 4. Configurable outbound extension support has been added to the outbound registration module. When set the extension will be placed in the Supported header. 5. Header parameters can now be configured on an outbound registration which will be placed in the Contact header. 6. Google specific OAuth / Bearer token authentication (draft-ietf-sipcore-sip-authn-02) has been added to the outbound registration module. All functionality changes are controlled by pjsip.conf configuration options and do not affect non-configured pjsip endpoints otherwise. ASTERISK-27971 #close Change-Id: Id214c2d1c550a41fcf564b7df8f3da7be565bd58
2018-07-18 12:45:26 +00:00
ast_sip_tpselector_unref(&selector);
return -1;
}
pj_list_insert_nodes_before(&route_set, route);
pjsip_regc_set_route_set(state->client_state->client, &route_set);
}
if (state->registration->line) {
ast_generate_random_string(state->client_state->line, sizeof(state->client_state->line));
}
pj_cstr(&server_uri, registration->server_uri);
if (sip_dialog_create_contact(pjsip_regc_get_pool(state->client_state->client),
&contact_uri, S_OR(registration->contact_user, "s"), &server_uri, &selector,
res_pjsip: Implement additional SIP RFCs for Google Voice trunk compatability This change implements a few different generic things which were brought on by Google Voice SIP. 1. The concept of flow transports have been introduced. These are configurable transports in pjsip.conf which can be used to reference a flow of signaling to a target. These have runtime configuration that can be changed by the signaling itself (such as Service-Routes and P-Preferred-Identity). When used these guarantee an individual connection (in the case of TCP or TLS) even if multiple flow transports exist to the same target. 2. Service-Routes (RFC 3608) support has been added to the outbound registration module which when received will be stored on the flow transport and used for requests referencing it. 3. P-Associated-URI / P-Preferred-Identity (RFC 3325) support has been added to the outbound registration module. If a P-Associated-URI header is received it will be used on requests as the P-Preferred-Identity. 4. Configurable outbound extension support has been added to the outbound registration module. When set the extension will be placed in the Supported header. 5. Header parameters can now be configured on an outbound registration which will be placed in the Contact header. 6. Google specific OAuth / Bearer token authentication (draft-ietf-sipcore-sip-authn-02) has been added to the outbound registration module. All functionality changes are controlled by pjsip.conf configuration options and do not affect non-configured pjsip endpoints otherwise. ASTERISK-27971 #close Change-Id: Id214c2d1c550a41fcf564b7df8f3da7be565bd58
2018-07-18 12:45:26 +00:00
state->client_state->line, registration->contact_header_params)) {
ast_sip_tpselector_unref(&selector);
return -1;
}
res_pjsip: Implement additional SIP RFCs for Google Voice trunk compatability This change implements a few different generic things which were brought on by Google Voice SIP. 1. The concept of flow transports have been introduced. These are configurable transports in pjsip.conf which can be used to reference a flow of signaling to a target. These have runtime configuration that can be changed by the signaling itself (such as Service-Routes and P-Preferred-Identity). When used these guarantee an individual connection (in the case of TCP or TLS) even if multiple flow transports exist to the same target. 2. Service-Routes (RFC 3608) support has been added to the outbound registration module which when received will be stored on the flow transport and used for requests referencing it. 3. P-Associated-URI / P-Preferred-Identity (RFC 3325) support has been added to the outbound registration module. If a P-Associated-URI header is received it will be used on requests as the P-Preferred-Identity. 4. Configurable outbound extension support has been added to the outbound registration module. When set the extension will be placed in the Supported header. 5. Header parameters can now be configured on an outbound registration which will be placed in the Contact header. 6. Google specific OAuth / Bearer token authentication (draft-ietf-sipcore-sip-authn-02) has been added to the outbound registration module. All functionality changes are controlled by pjsip.conf configuration options and do not affect non-configured pjsip endpoints otherwise. ASTERISK-27971 #close Change-Id: Id214c2d1c550a41fcf564b7df8f3da7be565bd58
2018-07-18 12:45:26 +00:00
ast_sip_tpselector_unref(&selector);
pj_cstr(&client_uri, registration->client_uri);
if (pjsip_regc_init(state->client_state->client, &server_uri, &client_uri,
&client_uri, 1, &contact_uri, registration->expiration) != PJ_SUCCESS) {
return -1;
}
return 0;
}
/*! \brief Helper function which performs a single registration */
static int sip_outbound_registration_perform(void *data)
{
struct sip_outbound_registration_state *state = data;
struct sip_outbound_registration *registration = ao2_bump(state->registration);
size_t i;
int max_delay;
/* Just in case the client state is being reused for this registration, free the auth information */
ast_sip_auth_vector_destroy(&state->client_state->outbound_auths);
ast_sip_security_mechanisms_vector_destroy(&state->client_state->security_mechanisms);
ast_sip_security_mechanisms_vector_destroy(&state->client_state->server_security_mechanisms);
AST_VECTOR_INIT(&state->client_state->outbound_auths, AST_VECTOR_SIZE(&registration->outbound_auths));
for (i = 0; i < AST_VECTOR_SIZE(&registration->outbound_auths); ++i) {
char *name = ast_strdup(AST_VECTOR_GET(&registration->outbound_auths, i));
if (name && AST_VECTOR_APPEND(&state->client_state->outbound_auths, name)) {
ast_free(name);
}
}
ast_sip_security_mechanisms_vector_copy(&state->client_state->security_mechanisms,
&registration->security_mechanisms);
state->client_state->retry_interval = registration->retry_interval;
state->client_state->forbidden_retry_interval = registration->forbidden_retry_interval;
state->client_state->fatal_retry_interval = registration->fatal_retry_interval;
state->client_state->max_retries = registration->max_retries;
state->client_state->retries = 0;
state->client_state->support_path = registration->support_path;
res_pjsip: Implement additional SIP RFCs for Google Voice trunk compatability This change implements a few different generic things which were brought on by Google Voice SIP. 1. The concept of flow transports have been introduced. These are configurable transports in pjsip.conf which can be used to reference a flow of signaling to a target. These have runtime configuration that can be changed by the signaling itself (such as Service-Routes and P-Preferred-Identity). When used these guarantee an individual connection (in the case of TCP or TLS) even if multiple flow transports exist to the same target. 2. Service-Routes (RFC 3608) support has been added to the outbound registration module which when received will be stored on the flow transport and used for requests referencing it. 3. P-Associated-URI / P-Preferred-Identity (RFC 3325) support has been added to the outbound registration module. If a P-Associated-URI header is received it will be used on requests as the P-Preferred-Identity. 4. Configurable outbound extension support has been added to the outbound registration module. When set the extension will be placed in the Supported header. 5. Header parameters can now be configured on an outbound registration which will be placed in the Contact header. 6. Google specific OAuth / Bearer token authentication (draft-ietf-sipcore-sip-authn-02) has been added to the outbound registration module. All functionality changes are controlled by pjsip.conf configuration options and do not affect non-configured pjsip endpoints otherwise. ASTERISK-27971 #close Change-Id: Id214c2d1c550a41fcf564b7df8f3da7be565bd58
2018-07-18 12:45:26 +00:00
state->client_state->support_outbound = registration->support_outbound;
state->client_state->security_negotiation = registration->security_negotiation;
state->client_state->auth_rejection_permanent = registration->auth_rejection_permanent;
max_delay = registration->max_random_initial_delay;
pjsip_regc_update_expires(state->client_state->client, registration->expiration);
/* n mod 0 is undefined, so don't let that happen */
schedule_registration(state->client_state, (max_delay ? ast_random() % max_delay : 0) + 1);
ao2_ref(registration, -1);
ao2_ref(state, -1);
return 0;
}
/*! \brief Apply function which finds or allocates a state structure */
static int sip_outbound_registration_apply(const struct ast_sorcery *sorcery, void *obj)
{
RAII_VAR(struct ao2_container *, states, ao2_global_obj_ref(current_states), ao2_cleanup);
res_pjsip_outbound_registration: Fix reload race condition. Performing a CLI "module reload" command when there are new pjsip.conf registration objects defined frequently failed to load them correctly. What happens is a race condition between res_pjsip pushing its reload into an asynchronous task processor task and the thread that does the rest of the reloads when it gets to reloading the res_pjsip_outbound_registration module. A similar race condition happens between a reload and the CLI/AMI show registrations commands. The reload updates the current_states container and the CLI/AMI commands call get_registrations() which builds a new current_states container. * Made res_pjsip.c reload_module() use ast_sip_push_task_synchronous() instead of ast_sip_push_task() to eliminate two threads processing config reloads at the same time. * Made get_registrations() not replace the global current_states container so the CLI/AMI show registrations command cannot interfere with reloading. You could never add/remove objects in the container without the possibility of the container being replaced out from under you by get_registrations(). * Added a registration loaded sorcery instance observer to purge any dead registration objects since get_registrations() cannot do this job anymore. The struct ast_sorcery_instance_observer callbacks must be used because the callback happens inline with the load process. The struct ast_sorcery_observer callbacks are pushed to a different thread. * Added some global current_states NULL pointer checks in case the container disappears because of unload_module(). * Made sorcery's struct ast_sorcery_instance_observer.object_type_loaded callbacks guaranteed to be called before any struct ast_sorcery_observer.loaded callbacks will be called. * Moved the check for non-reloadable objects to before the sorcery instance loading callbacks happen to short circuit unnecessary work. Previously with non-reloadable objects, the sorcery instance loading/loaded callbacks would always happen, the individual wizard loading/loaded would be prevented, and the non-reloadable type logging message would be logged for each associated wizard. ASTERISK-24729 #close Review: https://reviewboard.asterisk.org/r/4381/ ........ Merged revisions 431243 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431251 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-28 04:29:23 +00:00
RAII_VAR(struct sip_outbound_registration_state *, state, NULL, ao2_cleanup);
RAII_VAR(struct sip_outbound_registration_state *, new_state, NULL, ao2_cleanup);
struct sip_outbound_registration *applied = obj;
res_pjsip_outbound_registration: Fix reload race condition. Performing a CLI "module reload" command when there are new pjsip.conf registration objects defined frequently failed to load them correctly. What happens is a race condition between res_pjsip pushing its reload into an asynchronous task processor task and the thread that does the rest of the reloads when it gets to reloading the res_pjsip_outbound_registration module. A similar race condition happens between a reload and the CLI/AMI show registrations commands. The reload updates the current_states container and the CLI/AMI commands call get_registrations() which builds a new current_states container. * Made res_pjsip.c reload_module() use ast_sip_push_task_synchronous() instead of ast_sip_push_task() to eliminate two threads processing config reloads at the same time. * Made get_registrations() not replace the global current_states container so the CLI/AMI show registrations command cannot interfere with reloading. You could never add/remove objects in the container without the possibility of the container being replaced out from under you by get_registrations(). * Added a registration loaded sorcery instance observer to purge any dead registration objects since get_registrations() cannot do this job anymore. The struct ast_sorcery_instance_observer callbacks must be used because the callback happens inline with the load process. The struct ast_sorcery_observer callbacks are pushed to a different thread. * Added some global current_states NULL pointer checks in case the container disappears because of unload_module(). * Made sorcery's struct ast_sorcery_instance_observer.object_type_loaded callbacks guaranteed to be called before any struct ast_sorcery_observer.loaded callbacks will be called. * Moved the check for non-reloadable objects to before the sorcery instance loading callbacks happen to short circuit unnecessary work. Previously with non-reloadable objects, the sorcery instance loading/loaded callbacks would always happen, the individual wizard loading/loaded would be prevented, and the non-reloadable type logging message would be logged for each associated wizard. ASTERISK-24729 #close Review: https://reviewboard.asterisk.org/r/4381/ ........ Merged revisions 431243 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431251 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-28 04:29:23 +00:00
if (!states) {
/* Global container has gone. Likely shutting down. */
return -1;
}
state = ao2_find(states, ast_sorcery_object_get_id(applied), OBJ_SEARCH_KEY);
ast_debug(4, "Applying configuration to outbound registration '%s'\n", ast_sorcery_object_get_id(applied));
if (ast_strlen_zero(applied->server_uri)) {
ast_log(LOG_ERROR, "No server URI specified on outbound registration '%s'\n",
ast_sorcery_object_get_id(applied));
return -1;
} else if (ast_sip_validate_uri_length(applied->server_uri)) {
ast_log(LOG_ERROR, "Server URI or hostname length exceeds pjproject limit or is not a sip(s) uri: '%s'\n",
ast_sorcery_object_get_id(applied));
return -1;
} else if (ast_strlen_zero(applied->client_uri)) {
ast_log(LOG_ERROR, "No client URI specified on outbound registration '%s'\n",
ast_sorcery_object_get_id(applied));
return -1;
} else if (ast_sip_validate_uri_length(applied->client_uri)) {
ast_log(LOG_ERROR, "Client URI or hostname length exceeds pjproject limit or is not a sip(s) uri: '%s'\n",
ast_sorcery_object_get_id(applied));
return -1;
} else if (applied->line && ast_strlen_zero(applied->endpoint)) {
ast_log(LOG_ERROR, "Line support has been enabled on outbound registration '%s' without providing an endpoint\n",
ast_sorcery_object_get_id(applied));
return -1;
} else if (!ast_strlen_zero(applied->endpoint) && !applied->line) {
ast_log(LOG_ERROR, "An endpoint has been specified on outbound registration '%s' without enabling line support\n",
ast_sorcery_object_get_id(applied));
return -1;
}
if (state && can_reuse_registration(state->registration, applied)) {
ast_debug(4,
"No change between old configuration and new configuration on outbound registration '%s'. Using previous state\n",
ast_sorcery_object_get_id(applied));
res_pjsip_outbound_registration: Fix reload race condition. Performing a CLI "module reload" command when there are new pjsip.conf registration objects defined frequently failed to load them correctly. What happens is a race condition between res_pjsip pushing its reload into an asynchronous task processor task and the thread that does the rest of the reloads when it gets to reloading the res_pjsip_outbound_registration module. A similar race condition happens between a reload and the CLI/AMI show registrations commands. The reload updates the current_states container and the CLI/AMI commands call get_registrations() which builds a new current_states container. * Made res_pjsip.c reload_module() use ast_sip_push_task_synchronous() instead of ast_sip_push_task() to eliminate two threads processing config reloads at the same time. * Made get_registrations() not replace the global current_states container so the CLI/AMI show registrations command cannot interfere with reloading. You could never add/remove objects in the container without the possibility of the container being replaced out from under you by get_registrations(). * Added a registration loaded sorcery instance observer to purge any dead registration objects since get_registrations() cannot do this job anymore. The struct ast_sorcery_instance_observer callbacks must be used because the callback happens inline with the load process. The struct ast_sorcery_observer callbacks are pushed to a different thread. * Added some global current_states NULL pointer checks in case the container disappears because of unload_module(). * Made sorcery's struct ast_sorcery_instance_observer.object_type_loaded callbacks guaranteed to be called before any struct ast_sorcery_observer.loaded callbacks will be called. * Moved the check for non-reloadable objects to before the sorcery instance loading callbacks happen to short circuit unnecessary work. Previously with non-reloadable objects, the sorcery instance loading/loaded callbacks would always happen, the individual wizard loading/loaded would be prevented, and the non-reloadable type logging message would be logged for each associated wizard. ASTERISK-24729 #close Review: https://reviewboard.asterisk.org/r/4381/ ........ Merged revisions 431243 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431251 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-28 04:29:23 +00:00
/*
* This is OK to replace without relinking the state in the
* current_states container since state->registration and
* applied have the same key.
*/
ao2_lock(states);
ao2_replace(state->registration, applied);
res_pjsip_outbound_registration: Fix reload race condition. Performing a CLI "module reload" command when there are new pjsip.conf registration objects defined frequently failed to load them correctly. What happens is a race condition between res_pjsip pushing its reload into an asynchronous task processor task and the thread that does the rest of the reloads when it gets to reloading the res_pjsip_outbound_registration module. A similar race condition happens between a reload and the CLI/AMI show registrations commands. The reload updates the current_states container and the CLI/AMI commands call get_registrations() which builds a new current_states container. * Made res_pjsip.c reload_module() use ast_sip_push_task_synchronous() instead of ast_sip_push_task() to eliminate two threads processing config reloads at the same time. * Made get_registrations() not replace the global current_states container so the CLI/AMI show registrations command cannot interfere with reloading. You could never add/remove objects in the container without the possibility of the container being replaced out from under you by get_registrations(). * Added a registration loaded sorcery instance observer to purge any dead registration objects since get_registrations() cannot do this job anymore. The struct ast_sorcery_instance_observer callbacks must be used because the callback happens inline with the load process. The struct ast_sorcery_observer callbacks are pushed to a different thread. * Added some global current_states NULL pointer checks in case the container disappears because of unload_module(). * Made sorcery's struct ast_sorcery_instance_observer.object_type_loaded callbacks guaranteed to be called before any struct ast_sorcery_observer.loaded callbacks will be called. * Moved the check for non-reloadable objects to before the sorcery instance loading callbacks happen to short circuit unnecessary work. Previously with non-reloadable objects, the sorcery instance loading/loaded callbacks would always happen, the individual wizard loading/loaded would be prevented, and the non-reloadable type logging message would be logged for each associated wizard. ASTERISK-24729 #close Review: https://reviewboard.asterisk.org/r/4381/ ........ Merged revisions 431243 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431251 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-28 04:29:23 +00:00
ao2_unlock(states);
return 0;
}
if (!(new_state = sip_outbound_registration_state_alloc(applied))) {
return -1;
}
res_pjsip.c: Split ast_sip_push_task_synchronous() to fit expectations. ast_sip_push_task_synchronous() did not necessarily execute the passed in task under the specified serializer. If the current thread is any registered pjsip thread then it would execute the task immediately instead of under the specified serializer. Reentrancy issues could result if the task does not execute with the right serializer. The original reason ast_sip_push_task_synchronous() checked to see if the current thread was a registered pjsip thread was because of a deadlock with masquerades and the channel technology's fixup callback (ASTERISK_22936). A subsequent masquerade deadlock fix (ASTERISK_24356) involving call pickups avoided the original deadlock situation entirely. The PJSIP channel technology's fixup callback no longer needed to call ast_sip_push_task_synchronous(). However, there are a few places where this unexpected behavior is still required to avoid deadlocks. The pjsip monitor thread executes callbacks that do calls to ast_sip_push_task_synchronous() that would deadlock if the task were actually pushed to the specified serializer. I ran into one dealing with the pubsub subscriptions where an ao2 destructor called ast_sip_push_task_synchronous(). * Split ast_sip_push_task_synchronous() into ast_sip_push_task_wait_servant() and ast_sip_push_task_wait_serializer(). ast_sip_push_task_wait_servant() has the old behavior of ast_sip_push_task_synchronous(). ast_sip_push_task_wait_serializer() has the new behavior where the task is always executed by the specified serializer or a picked serializer if one is not passed in. Both functions behave the same if the current thread is not a SIP servant. * Redirected ast_sip_push_task_synchronous() to ast_sip_push_task_wait_servant() to preserve API for released branches. ASTERISK_26806 Change-Id: Id040fa42c0e5972f4c8deef380921461d213b9f3
2018-03-27 16:04:42 +00:00
if (ast_sip_push_task_wait_serializer(new_state->client_state->serializer,
sip_outbound_registration_regc_alloc, new_state)) {
return -1;
}
if (ast_sip_push_task(new_state->client_state->serializer,
sip_outbound_registration_perform, ao2_bump(new_state))) {
ast_log(LOG_ERROR, "Failed to perform outbound registration on '%s'\n",
ast_sorcery_object_get_id(new_state->registration));
ao2_ref(new_state, -1);
return -1;
}
ao2_lock(states);
if (state) {
ao2_unlink(states, state);
}
ao2_link(states, new_state);
ao2_unlock(states);
return 0;
}
static int security_mechanism_to_str(const void *obj, const intptr_t *args, char **buf)
{
const struct sip_outbound_registration *registration = obj;
return ast_sip_security_mechanisms_to_str(&registration->security_mechanisms, 0, buf);
}
static const char *security_negotiation_map[] = {
[AST_SIP_SECURITY_NEG_NONE] = "no",
[AST_SIP_SECURITY_NEG_MEDIASEC] = "mediasec",
};
static int security_negotiation_to_str(const void *obj, const intptr_t *args, char **buf)
{
const struct sip_outbound_registration *registration = obj;
if (ARRAY_IN_BOUNDS(registration->security_negotiation, security_negotiation_map)) {
*buf = ast_strdup(security_negotiation_map[registration->security_negotiation]);
}
return 0;
}
static int security_mechanisms_handler(const struct aco_option *opt, struct ast_variable *var, void *obj)
{
struct sip_outbound_registration *registration = obj;
return ast_sip_security_mechanism_vector_init(&registration->security_mechanisms, var->value);
}
static int security_negotiation_handler(const struct aco_option *opt, struct ast_variable *var, void *obj)
{
struct sip_outbound_registration *registration = obj;
return ast_sip_set_security_negotiation(&registration->security_negotiation, var->value);
}
static int outbound_auth_handler(const struct aco_option *opt, struct ast_variable *var, void *obj)
{
struct sip_outbound_registration *registration = obj;
return ast_sip_auth_vector_init(&registration->outbound_auths, var->value);
}
res_pjsip: AMI commands and events. 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/ ........ Merged revisions 403131 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403133 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-11-23 17:26:57 +00:00
static int outbound_auths_to_str(const void *obj, const intptr_t *args, char **buf)
{
const struct sip_outbound_registration *registration = obj;
sorcery: Create AST_SORCERY dialplan function. 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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410042 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-06 22:39:54 +00:00
res_pjsip: AMI commands and events. 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/ ........ Merged revisions 403131 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403133 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-11-23 17:26:57 +00:00
return ast_sip_auths_to_str(&registration->outbound_auths, buf);
}
sorcery: Create AST_SORCERY dialplan function. 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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410042 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-06 22:39:54 +00:00
static int outbound_auths_to_var_list(const void *obj, struct ast_variable **fields)
{
const struct sip_outbound_registration *registration = obj;
int i;
struct ast_variable *head = NULL;
for (i = 0; i < AST_VECTOR_SIZE(&registration->outbound_auths) ; i++) {
ast_variable_list_append(&head, ast_variable_new("outbound_auth",
AST_VECTOR_GET(&registration->outbound_auths, i), ""));
}
if (head) {
*fields = head;
}
return 0;
}
static int unregister_task(void *obj)
{
struct sip_outbound_registration_state *state = obj;
struct pjsip_regc *client = state->client_state->client;
pjsip_tx_data *tdata;
pjsip_regc_info info;
pjsip_regc_get_info(client, &info);
ast_debug(1, "Unregistering contacts with server '%s' from client '%s'\n",
state->registration->server_uri, state->registration->client_uri);
cancel_registration(state->client_state);
res_pjsip: Implement additional SIP RFCs for Google Voice trunk compatability This change implements a few different generic things which were brought on by Google Voice SIP. 1. The concept of flow transports have been introduced. These are configurable transports in pjsip.conf which can be used to reference a flow of signaling to a target. These have runtime configuration that can be changed by the signaling itself (such as Service-Routes and P-Preferred-Identity). When used these guarantee an individual connection (in the case of TCP or TLS) even if multiple flow transports exist to the same target. 2. Service-Routes (RFC 3608) support has been added to the outbound registration module which when received will be stored on the flow transport and used for requests referencing it. 3. P-Associated-URI / P-Preferred-Identity (RFC 3325) support has been added to the outbound registration module. If a P-Associated-URI header is received it will be used on requests as the P-Preferred-Identity. 4. Configurable outbound extension support has been added to the outbound registration module. When set the extension will be placed in the Supported header. 5. Header parameters can now be configured on an outbound registration which will be placed in the Contact header. 6. Google specific OAuth / Bearer token authentication (draft-ietf-sipcore-sip-authn-02) has been added to the outbound registration module. All functionality changes are controlled by pjsip.conf configuration options and do not affect non-configured pjsip endpoints otherwise. ASTERISK-27971 #close Change-Id: Id214c2d1c550a41fcf564b7df8f3da7be565bd58
2018-07-18 12:45:26 +00:00
if (pjsip_regc_unregister(client, &tdata) == PJ_SUCCESS
&& add_configured_supported_headers(state->client_state, tdata)) {
registration_client_send(state->client_state, tdata);
}
ao2_ref(state, -1);
return 0;
}
static int queue_unregister(struct sip_outbound_registration_state *state)
{
ao2_ref(state, +1);
if (ast_sip_push_task(state->client_state->serializer, unregister_task, state)) {
ao2_ref(state, -1);
return -1;
}
return 0;
}
static int queue_register(struct sip_outbound_registration_state *state)
{
ao2_ref(state, +1);
if (ast_sip_push_task(state->client_state->serializer, sip_outbound_registration_perform, state)) {
ao2_ref(state, -1);
return -1;
}
return 0;
}
static void unregister_all(void)
{
struct ao2_container *states;
states = ao2_global_obj_ref(current_states);
if (!states) {
return;
}
/* Clean out all the states and let sorcery handle recreating the registrations */
ao2_callback(states, OBJ_UNLINK | OBJ_NODATA | OBJ_MULTIPLE, NULL, NULL);
ao2_ref(states, -1);
}
static void reregister_all(void)
{
unregister_all();
ast_sorcery_load_object(ast_sip_get_sorcery(), "registration");
}
static char *cli_complete_registration(const char *line, const char *word,
int pos, int state)
{
char *result = NULL;
int wordlen;
int which = 0;
struct sip_outbound_registration *registration;
struct ao2_container *registrations;
struct ao2_iterator i;
if (pos != 3) {
return NULL;
}
wordlen = strlen(word);
if (wordlen == 0 && ++which > state) {
return ast_strdup("*all");
}
registrations = ast_sorcery_retrieve_by_fields(ast_sip_get_sorcery(), "registration",
AST_RETRIEVE_FLAG_MULTIPLE | AST_RETRIEVE_FLAG_ALL, NULL);
if (!registrations) {
return NULL;
}
i = ao2_iterator_init(registrations, 0);
while ((registration = ao2_iterator_next(&i))) {
const char *name = ast_sorcery_object_get_id(registration);
if (!strncasecmp(word, name, wordlen) && ++which > state) {
result = ast_strdup(name);
}
ao2_ref(registration, -1);
if (result) {
break;
}
}
ao2_iterator_destroy(&i);
ao2_ref(registrations, -1);
return result;
}
static char *cli_unregister(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
struct sip_outbound_registration_state *state;
const char *registration_name;
switch (cmd) {
case CLI_INIT:
e->command = "pjsip send unregister";
e->usage =
"Usage: pjsip send unregister <registration> | *all\n"
" Unregisters the specified (or all) outbound registration(s) "
"and stops future registration attempts.\n";
return NULL;
case CLI_GENERATE:
return cli_complete_registration(a->line, a->word, a->pos, a->n);
}
if (a->argc != 4) {
return CLI_SHOWUSAGE;
}
registration_name = a->argv[3];
if (strcmp(registration_name, "*all") == 0) {
unregister_all();
ast_cli(a->fd, "Unregister all queued\n");
return CLI_SUCCESS;
}
state = get_state(registration_name);
if (!state) {
ast_cli(a->fd, "Unable to retrieve registration %s\n", registration_name);
return CLI_FAILURE;
}
if (queue_unregister(state)) {
ast_cli(a->fd, "Failed to queue unregistration\n");
}
ao2_ref(state, -1);
return CLI_SUCCESS;
}
static char *cli_register(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
struct sip_outbound_registration_state *state;
const char *registration_name;
switch (cmd) {
case CLI_INIT:
e->command = "pjsip send register";
e->usage =
"Usage: pjsip send register <registration> | *all \n"
" Unregisters the specified (or all) outbound "
"registration(s) then starts registration(s) and schedules re-registrations.\n";
return NULL;
case CLI_GENERATE:
return cli_complete_registration(a->line, a->word, a->pos, a->n);
}
if (a->argc != 4) {
return CLI_SHOWUSAGE;
}
registration_name = a->argv[3];
if (strcmp(registration_name, "*all") == 0) {
reregister_all();
ast_cli(a->fd, "Re-register all queued\n");
return CLI_SUCCESS;
}
state = get_state(registration_name);
if (!state) {
ast_cli(a->fd, "Unable to retrieve registration %s\n", registration_name);
return CLI_FAILURE;
}
/* We need to serialize the unregister and register so they need
* to be queued as separate tasks.
*/
if (queue_unregister(state)) {
ast_cli(a->fd, "Failed to queue unregistration\n");
} else if (queue_register(state)) {
ast_cli(a->fd, "Failed to queue registration\n");
}
ao2_ref(state, -1);
return CLI_SUCCESS;
}
static int ami_unregister(struct mansession *s, const struct message *m)
{
const char *registration_name = astman_get_header(m, "Registration");
struct sip_outbound_registration_state *state;
if (ast_strlen_zero(registration_name)) {
astman_send_error(s, m, "Registration parameter missing.");
return 0;
}
if (strcmp(registration_name, "*all") == 0) {
unregister_all();
astman_send_ack(s, m, "Unregistrations queued.");
return 0;
}
state = get_state(registration_name);
if (!state) {
astman_send_error(s, m, "Unable to retrieve registration entry\n");
return 0;
}
if (queue_unregister(state)) {
astman_send_ack(s, m, "Failed to queue unregistration");
} else {
astman_send_ack(s, m, "Unregistration sent");
}
ao2_ref(state, -1);
return 0;
}
static int ami_register(struct mansession *s, const struct message *m)
{
const char *registration_name = astman_get_header(m, "Registration");
struct sip_outbound_registration_state *state;
if (ast_strlen_zero(registration_name)) {
astman_send_error(s, m, "Registration parameter missing.");
return 0;
}
if (strcmp(registration_name, "*all") == 0) {
reregister_all();
astman_send_ack(s, m, "Reregistrations queued.");
return 0;
}
state = get_state(registration_name);
if (!state) {
astman_send_error(s, m, "Unable to retrieve registration entry\n");
return 0;
}
/* We need to serialize the unregister and register so they need
* to be queued as separate tasks.
*/
if (queue_unregister(state)) {
astman_send_ack(s, m, "Failed to queue unregistration");
} else if (queue_register(state)) {
astman_send_ack(s, m, "Failed to queue unregistration");
} else {
astman_send_ack(s, m, "Reregistration sent");
}
ao2_ref(state, -1);
return 0;
}
res_pjsip: AMI commands and events. 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/ ........ Merged revisions 403131 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403133 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-11-23 17:26:57 +00:00
struct sip_ami_outbound {
struct ast_sip_ami *ami;
int registered;
int not_registered;
struct sip_outbound_registration *registration;
};
static int ami_outbound_registration_task(void *obj)
{
struct sip_ami_outbound *ami = obj;
struct ast_str *buf;
struct sip_outbound_registration_state *state;
res_pjsip: AMI commands and events. 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/ ........ Merged revisions 403131 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403133 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-11-23 17:26:57 +00:00
AMI: Make AMI actions that generate event lists consistent. * 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/ ........ 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
2015-01-09 18:16:54 +00:00
buf = ast_sip_create_ami_event("OutboundRegistrationDetail", ami->ami);
res_pjsip: AMI commands and events. 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/ ........ Merged revisions 403131 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403133 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-11-23 17:26:57 +00:00
if (!buf) {
return -1;
}
ast_sip_sorcery_object_to_ami(ami->registration, &buf);
if ((state = get_state(ast_sorcery_object_get_id(ami->registration)))) {
pjsip_regc_info info;
AMI: Make AMI actions that generate event lists consistent. * 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/ ........ 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
2015-01-09 18:16:54 +00:00
if (state->client_state->status == SIP_REGISTRATION_REGISTERED) {
res_pjsip: AMI commands and events. 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/ ........ Merged revisions 403131 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403133 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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++ami->registered;
} else {
++ami->not_registered;
}
AMI: Make AMI actions that generate event lists consistent. * 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/ ........ 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
2015-01-09 18:16:54 +00:00
ast_str_append(&buf, 0, "Status: %s\r\n",
res_pjsip_outbound_registration.c: Fix handle_client_state_destruction() refs * handle_client_state_destruction() must always be passed a ref to client_state because it will always unref client_state. handle_registration_response() was not passing a client_state ref. * Made the final un-REGISTER message get sent normally using the pjproject register control structure in handle_client_state_destruction(). The previous code attempted to short circuit the response handling for the module to unload. That doesn't work for a couple reasons. One, pjsip_regc_send() may call the registered callback before it returns and unbalance the client_state ref count. Two, the registered callback handles any authentication for the un-REGISTER message. * Made the distinction between internal registration state and external registration status with sip_outbound_registration_status_str(). This is necessary to avoid altering documented AMI messages with internal changes. * Removed references to client_state->client outside of the serializer thread. When handle_client_state_destruction() destroys the pjproject register control structure that memory is freed and cannot be referenced anymore. These accesses were to provide information for debug and off-nominal warning messages. * In sip_outbound_registration_timer_cb() you should not access entry->id after unrefing client_state because the passed in entry is normally pointing to the timer entry in the client_state object. ASTERISK-24907 Reported by: Kevin Harwell Change-Id: Ia7b446d8644b6b4550ef5bea49527671de65183f
2015-06-16 20:06:22 +00:00
sip_outbound_registration_status_str(state->client_state->status));
res_pjsip: AMI commands and events. 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/ ........ Merged revisions 403131 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403133 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-11-23 17:26:57 +00:00
pjsip_regc_get_info(state->client_state->client, &info);
ast_str_append(&buf, 0, "NextReg: %d\r\n", info.next_reg);
ao2_ref(state, -1);
res_pjsip: AMI commands and events. 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/ ........ Merged revisions 403131 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403133 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-11-23 17:26:57 +00:00
}
astman_append(ami->ami->s, "%s\r\n", ast_str_buffer(buf));
ast_free(buf);
res_pjsip: AMI commands and events. 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/ ........ Merged revisions 403131 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403133 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-11-23 17:26:57 +00:00
return ast_sip_format_auths_ami(&ami->registration->outbound_auths, ami->ami);
}
static int ami_outbound_registration_detail(void *obj, void *arg, int flags)
{
struct sip_ami_outbound *ami = arg;
ami->registration = obj;
res_pjsip.c: Split ast_sip_push_task_synchronous() to fit expectations. ast_sip_push_task_synchronous() did not necessarily execute the passed in task under the specified serializer. If the current thread is any registered pjsip thread then it would execute the task immediately instead of under the specified serializer. Reentrancy issues could result if the task does not execute with the right serializer. The original reason ast_sip_push_task_synchronous() checked to see if the current thread was a registered pjsip thread was because of a deadlock with masquerades and the channel technology's fixup callback (ASTERISK_22936). A subsequent masquerade deadlock fix (ASTERISK_24356) involving call pickups avoided the original deadlock situation entirely. The PJSIP channel technology's fixup callback no longer needed to call ast_sip_push_task_synchronous(). However, there are a few places where this unexpected behavior is still required to avoid deadlocks. The pjsip monitor thread executes callbacks that do calls to ast_sip_push_task_synchronous() that would deadlock if the task were actually pushed to the specified serializer. I ran into one dealing with the pubsub subscriptions where an ao2 destructor called ast_sip_push_task_synchronous(). * Split ast_sip_push_task_synchronous() into ast_sip_push_task_wait_servant() and ast_sip_push_task_wait_serializer(). ast_sip_push_task_wait_servant() has the old behavior of ast_sip_push_task_synchronous(). ast_sip_push_task_wait_serializer() has the new behavior where the task is always executed by the specified serializer or a picked serializer if one is not passed in. Both functions behave the same if the current thread is not a SIP servant. * Redirected ast_sip_push_task_synchronous() to ast_sip_push_task_wait_servant() to preserve API for released branches. ASTERISK_26806 Change-Id: Id040fa42c0e5972f4c8deef380921461d213b9f3
2018-03-27 16:04:42 +00:00
return ast_sip_push_task_wait_servant(NULL, ami_outbound_registration_task, ami);
res_pjsip: AMI commands and events. 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/ ........ Merged revisions 403131 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403133 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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}
static int ami_show_outbound_registrations(struct mansession *s,
const struct message *m)
{
struct ast_sip_ami ami = { .s = s, .m = m, .action_id = astman_get_header(m, "ActionID"), };
res_pjsip: AMI commands and events. 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/ ........ Merged revisions 403131 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403133 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-11-23 17:26:57 +00:00
struct sip_ami_outbound ami_outbound = { .ami = &ami };
struct ao2_container *regs;
res_pjsip: AMI commands and events. 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/ ........ Merged revisions 403131 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403133 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-11-23 17:26:57 +00:00
regs = get_registrations();
res_pjsip: AMI commands and events. 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/ ........ Merged revisions 403131 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403133 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-11-23 17:26:57 +00:00
if (!regs) {
astman_send_error(s, m, "Unable to retrieve "
res_pjsip: AMI commands and events. 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/ ........ Merged revisions 403131 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403133 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-11-23 17:26:57 +00:00
"outbound registrations\n");
return -1;
}
astman_send_listack(s, m, "Following are Events for each Outbound registration",
"start");
res_pjsip: AMI commands and events. 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/ ........ Merged revisions 403131 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403133 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-11-23 17:26:57 +00:00
ao2_callback(regs, OBJ_NODATA, ami_outbound_registration_detail, &ami_outbound);
AMI: Make AMI actions that generate event lists consistent. * 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/ ........ 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
2015-01-09 18:16:54 +00:00
astman_send_list_complete_start(s, m, "OutboundRegistrationDetailComplete",
ami_outbound.registered + ami_outbound.not_registered);
astman_append(s,
"Registered: %d\r\n"
"NotRegistered: %d\r\n",
ami_outbound.registered,
ami_outbound.not_registered);
astman_send_list_complete_end(s);
ao2_ref(regs, -1);
res_pjsip: AMI commands and events. 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/ ........ Merged revisions 403131 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403133 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-11-23 17:26:57 +00:00
return 0;
}
static struct ao2_container *cli_get_container(const char *regex)
{
RAII_VAR(struct ao2_container *, container, NULL, ao2_cleanup);
struct ao2_container *s_container;
container = ast_sorcery_retrieve_by_regex(ast_sip_get_sorcery(), "registration", regex);
if (!container) {
return NULL;
}
s_container = ao2_container_alloc_list(AO2_ALLOC_OPT_LOCK_NOLOCK, 0,
ast_sorcery_object_id_sort, ast_sorcery_object_id_compare);
if (!s_container) {
return NULL;
}
if (ao2_container_dup(s_container, container, 0)) {
ao2_ref(s_container, -1);
return NULL;
}
return s_container;
}
static int cli_iterator(void *container, ao2_callback_fn callback, void *args)
{
ao2_callback(container, OBJ_NODATA, callback, args);
return 0;
}
static void *cli_retrieve_by_id(const char *id)
{
struct ao2_container *states;
void *obj = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "registration", id);
if (!obj) {
/* if the object no longer exists then remove its state */
res_pjsip_outbound_registration: Fix reload race condition. Performing a CLI "module reload" command when there are new pjsip.conf registration objects defined frequently failed to load them correctly. What happens is a race condition between res_pjsip pushing its reload into an asynchronous task processor task and the thread that does the rest of the reloads when it gets to reloading the res_pjsip_outbound_registration module. A similar race condition happens between a reload and the CLI/AMI show registrations commands. The reload updates the current_states container and the CLI/AMI commands call get_registrations() which builds a new current_states container. * Made res_pjsip.c reload_module() use ast_sip_push_task_synchronous() instead of ast_sip_push_task() to eliminate two threads processing config reloads at the same time. * Made get_registrations() not replace the global current_states container so the CLI/AMI show registrations command cannot interfere with reloading. You could never add/remove objects in the container without the possibility of the container being replaced out from under you by get_registrations(). * Added a registration loaded sorcery instance observer to purge any dead registration objects since get_registrations() cannot do this job anymore. The struct ast_sorcery_instance_observer callbacks must be used because the callback happens inline with the load process. The struct ast_sorcery_observer callbacks are pushed to a different thread. * Added some global current_states NULL pointer checks in case the container disappears because of unload_module(). * Made sorcery's struct ast_sorcery_instance_observer.object_type_loaded callbacks guaranteed to be called before any struct ast_sorcery_observer.loaded callbacks will be called. * Moved the check for non-reloadable objects to before the sorcery instance loading callbacks happen to short circuit unnecessary work. Previously with non-reloadable objects, the sorcery instance loading/loaded callbacks would always happen, the individual wizard loading/loaded would be prevented, and the non-reloadable type logging message would be logged for each associated wizard. ASTERISK-24729 #close Review: https://reviewboard.asterisk.org/r/4381/ ........ Merged revisions 431243 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431251 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-28 04:29:23 +00:00
states = ao2_global_obj_ref(current_states);
if (states) {
ao2_find(states, id, OBJ_SEARCH_KEY | OBJ_UNLINK | OBJ_NODATA);
ao2_ref(states, -1);
}
}
return obj;
}
static int cli_print_header(void *obj, void *arg, int flags)
{
struct ast_sip_cli_context *context = arg;
ast_assert(context->output_buffer != NULL);
ast_str_append(&context->output_buffer, 0,
" <Registration/ServerURI..............................> <Auth....................> <Status.......>\n");
return 0;
}
static int cli_print_body(void *obj, void *arg, int flags)
{
struct sip_outbound_registration *registration = obj;
struct ast_sip_cli_context *context = arg;
const char *id = ast_sorcery_object_get_id(registration);
struct sip_outbound_registration_state *state = get_state(id);
int expsecs;
#define REGISTRATION_URI_FIELD_LEN 53
ast_assert(context->output_buffer != NULL);
expsecs = state ? state->client_state->registration_expires - ((int) time(NULL)) : 0;
ast_str_append(&context->output_buffer, 0, " %-s/%-*.*s %-26s %-16s %s%d%s\n",
id,
(int) (REGISTRATION_URI_FIELD_LEN - strlen(id)),
(int) (REGISTRATION_URI_FIELD_LEN - strlen(id)),
registration->server_uri,
AST_VECTOR_SIZE(&registration->outbound_auths)
? AST_VECTOR_GET(&registration->outbound_auths, 0)
: "n/a",
(state ? sip_outbound_registration_status_str(state->client_state->status) : "Unregistered"),
state ? " (exp. " : "", abs(expsecs), state ? (expsecs < 0 ? "s ago)" : "s)") : "");
ao2_cleanup(state);
if (context->show_details
|| (context->show_details_only_level_0 && context->indent_level == 0)) {
ast_str_append(&context->output_buffer, 0, "\n");
ast_sip_cli_print_sorcery_objectset(registration, context, 0);
}
return 0;
}
/*
* A function pointer to callback needs to be within the
* module in order to avoid problems with an undefined
* symbol when the module is loaded.
*/
static char *my_cli_traverse_objects(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
return ast_sip_cli_traverse_objects(e, cmd, a);
}
static struct ast_cli_entry cli_outbound_registration[] = {
AST_CLI_DEFINE(cli_unregister, "Unregisters outbound registration target"),
AST_CLI_DEFINE(cli_register, "Registers an outbound registration target"),
AST_CLI_DEFINE(my_cli_traverse_objects, "List PJSIP Registrations",
.command = "pjsip list registrations",
.usage = "Usage: pjsip list registrations [ like <pattern> ]\n"
" List the configured PJSIP Registrations\n"
" Optional regular expression pattern is used to filter the list.\n"),
AST_CLI_DEFINE(my_cli_traverse_objects, "Show PJSIP Registrations",
.command = "pjsip show registrations",
.usage = "Usage: pjsip show registrations [ like <pattern> ]\n"
" Show the configured PJSIP Registrations\n"
" Optional regular expression pattern is used to filter the list.\n"),
AST_CLI_DEFINE(my_cli_traverse_objects, "Show PJSIP Registration",
.command = "pjsip show registration",
.usage = "Usage: pjsip show registration <id>\n"
" Show the configured PJSIP Registration\n"),
};
static struct ast_sip_cli_formatter_entry *cli_formatter;
static void auth_observer(const char *type)
{
struct sip_outbound_registration *registration;
struct sip_outbound_registration_state *state;
struct ao2_container *regs;
const char *registration_id;
struct ao2_iterator i;
ast_debug(4, "Auths updated. Checking for any outbound registrations that are in permanent rejected state so they can be retried\n");
regs = ast_sorcery_retrieve_by_fields(ast_sip_get_sorcery(), "registration",
AST_RETRIEVE_FLAG_MULTIPLE | AST_RETRIEVE_FLAG_ALL, NULL);
if (!regs || ao2_container_count(regs) == 0) {
ao2_cleanup(regs);
return;
}
i = ao2_iterator_init(regs, 0);
for (; (registration = ao2_iterator_next(&i)); ao2_ref(registration, -1)) {
registration_id = ast_sorcery_object_get_id(registration);
state = get_state(registration_id);
if (state && state->client_state->status == SIP_REGISTRATION_REJECTED_PERMANENT) {
ast_debug(4, "Trying outbound registration '%s' again\n", registration_id);
if (ast_sip_push_task(state->client_state->serializer,
sip_outbound_registration_perform, ao2_bump(state))) {
ast_log(LOG_ERROR, "Failed to perform outbound registration on '%s'\n", registration_id);
ao2_ref(state, -1);
}
}
ao2_cleanup(state);
}
ao2_iterator_destroy(&i);
ao2_cleanup(regs);
}
res_pjsip_outbound_registration: Fix reload race condition. Performing a CLI "module reload" command when there are new pjsip.conf registration objects defined frequently failed to load them correctly. What happens is a race condition between res_pjsip pushing its reload into an asynchronous task processor task and the thread that does the rest of the reloads when it gets to reloading the res_pjsip_outbound_registration module. A similar race condition happens between a reload and the CLI/AMI show registrations commands. The reload updates the current_states container and the CLI/AMI commands call get_registrations() which builds a new current_states container. * Made res_pjsip.c reload_module() use ast_sip_push_task_synchronous() instead of ast_sip_push_task() to eliminate two threads processing config reloads at the same time. * Made get_registrations() not replace the global current_states container so the CLI/AMI show registrations command cannot interfere with reloading. You could never add/remove objects in the container without the possibility of the container being replaced out from under you by get_registrations(). * Added a registration loaded sorcery instance observer to purge any dead registration objects since get_registrations() cannot do this job anymore. The struct ast_sorcery_instance_observer callbacks must be used because the callback happens inline with the load process. The struct ast_sorcery_observer callbacks are pushed to a different thread. * Added some global current_states NULL pointer checks in case the container disappears because of unload_module(). * Made sorcery's struct ast_sorcery_instance_observer.object_type_loaded callbacks guaranteed to be called before any struct ast_sorcery_observer.loaded callbacks will be called. * Moved the check for non-reloadable objects to before the sorcery instance loading callbacks happen to short circuit unnecessary work. Previously with non-reloadable objects, the sorcery instance loading/loaded callbacks would always happen, the individual wizard loading/loaded would be prevented, and the non-reloadable type logging message would be logged for each associated wizard. ASTERISK-24729 #close Review: https://reviewboard.asterisk.org/r/4381/ ........ Merged revisions 431243 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431251 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-28 04:29:23 +00:00
static const struct ast_sorcery_observer observer_callbacks_auth = {
.loaded = auth_observer,
};
res_pjsip_outbound_registration: Fix reload race condition. Performing a CLI "module reload" command when there are new pjsip.conf registration objects defined frequently failed to load them correctly. What happens is a race condition between res_pjsip pushing its reload into an asynchronous task processor task and the thread that does the rest of the reloads when it gets to reloading the res_pjsip_outbound_registration module. A similar race condition happens between a reload and the CLI/AMI show registrations commands. The reload updates the current_states container and the CLI/AMI commands call get_registrations() which builds a new current_states container. * Made res_pjsip.c reload_module() use ast_sip_push_task_synchronous() instead of ast_sip_push_task() to eliminate two threads processing config reloads at the same time. * Made get_registrations() not replace the global current_states container so the CLI/AMI show registrations command cannot interfere with reloading. You could never add/remove objects in the container without the possibility of the container being replaced out from under you by get_registrations(). * Added a registration loaded sorcery instance observer to purge any dead registration objects since get_registrations() cannot do this job anymore. The struct ast_sorcery_instance_observer callbacks must be used because the callback happens inline with the load process. The struct ast_sorcery_observer callbacks are pushed to a different thread. * Added some global current_states NULL pointer checks in case the container disappears because of unload_module(). * Made sorcery's struct ast_sorcery_instance_observer.object_type_loaded callbacks guaranteed to be called before any struct ast_sorcery_observer.loaded callbacks will be called. * Moved the check for non-reloadable objects to before the sorcery instance loading callbacks happen to short circuit unnecessary work. Previously with non-reloadable objects, the sorcery instance loading/loaded callbacks would always happen, the individual wizard loading/loaded would be prevented, and the non-reloadable type logging message would be logged for each associated wizard. ASTERISK-24729 #close Review: https://reviewboard.asterisk.org/r/4381/ ........ Merged revisions 431243 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431251 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-28 04:29:23 +00:00
static int check_state(void *obj, void *arg, int flags)
{
struct sip_outbound_registration_state *state = obj;
struct sip_outbound_registration *registration;
registration = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "registration",
ast_sorcery_object_get_id(state->registration));
if (!registration) {
/* This is a dead registration */
return CMP_MATCH;
}
ao2_ref(registration, -1);
return 0;
}
/*!
* \internal
* \brief Observer to purge dead registration states.
*
* \param name Module name owning the sorcery instance.
* \param sorcery Instance being observed.
* \param object_type Name of object being observed.
* \param reloaded Non-zero if the object is being reloaded.
*/
static void registration_loaded_observer(const char *name, const struct ast_sorcery *sorcery, const char *object_type, int reloaded)
{
struct ao2_container *states;
if (strcmp(object_type, "registration")) {
/* Not interested */
return;
}
states = ao2_global_obj_ref(current_states);
if (!states) {
/* Global container has gone. Likely shutting down. */
return;
}
/*
* Refresh the current configured registrations. We don't need to hold
* onto the objects, as the apply handler will cause their states to
* be created appropriately.
*/
ao2_cleanup(get_registrations());
res_pjsip_outbound_registration: Fix reload race condition. Performing a CLI "module reload" command when there are new pjsip.conf registration objects defined frequently failed to load them correctly. What happens is a race condition between res_pjsip pushing its reload into an asynchronous task processor task and the thread that does the rest of the reloads when it gets to reloading the res_pjsip_outbound_registration module. A similar race condition happens between a reload and the CLI/AMI show registrations commands. The reload updates the current_states container and the CLI/AMI commands call get_registrations() which builds a new current_states container. * Made res_pjsip.c reload_module() use ast_sip_push_task_synchronous() instead of ast_sip_push_task() to eliminate two threads processing config reloads at the same time. * Made get_registrations() not replace the global current_states container so the CLI/AMI show registrations command cannot interfere with reloading. You could never add/remove objects in the container without the possibility of the container being replaced out from under you by get_registrations(). * Added a registration loaded sorcery instance observer to purge any dead registration objects since get_registrations() cannot do this job anymore. The struct ast_sorcery_instance_observer callbacks must be used because the callback happens inline with the load process. The struct ast_sorcery_observer callbacks are pushed to a different thread. * Added some global current_states NULL pointer checks in case the container disappears because of unload_module(). * Made sorcery's struct ast_sorcery_instance_observer.object_type_loaded callbacks guaranteed to be called before any struct ast_sorcery_observer.loaded callbacks will be called. * Moved the check for non-reloadable objects to before the sorcery instance loading callbacks happen to short circuit unnecessary work. Previously with non-reloadable objects, the sorcery instance loading/loaded callbacks would always happen, the individual wizard loading/loaded would be prevented, and the non-reloadable type logging message would be logged for each associated wizard. ASTERISK-24729 #close Review: https://reviewboard.asterisk.org/r/4381/ ........ Merged revisions 431243 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431251 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-28 04:29:23 +00:00
/* Now to purge dead registrations. */
ao2_callback(states, OBJ_UNLINK | OBJ_NODATA | OBJ_MULTIPLE, check_state, NULL);
ao2_ref(states, -1);
}
static const struct ast_sorcery_instance_observer observer_callbacks_registrations = {
.object_type_loaded = registration_loaded_observer,
};
static void registration_deleted_observer(const void *obj)
{
const struct sip_outbound_registration *registration = obj;
struct ao2_container *states;
states = ao2_global_obj_ref(current_states);
if (!states) {
/* Global container has gone. Likely shutting down. */
return;
}
ao2_find(states, ast_sorcery_object_get_id(registration), OBJ_UNLINK | OBJ_NODATA | OBJ_SEARCH_KEY);
ao2_ref(states, -1);
}
static const struct ast_sorcery_observer registration_observer = {
.deleted = registration_deleted_observer,
};
static void network_change_stasis_cb(void *data, struct stasis_subscription *sub, struct stasis_message *message)
{
/* This callback is only concerned with network change messages from the system topic. */
if (stasis_message_type(message) != ast_network_change_type()) {
return;
}
ast_debug(3, "Received network change event\n");
reregister_all();
}
static int unload_module(void)
{
int remaining;
network_change_sub = stasis_unsubscribe_and_join(network_change_sub);
ast_manager_unregister("PJSIPShowRegistrationsOutbound");
ast_manager_unregister("PJSIPUnregister");
ast_manager_unregister("PJSIPRegister");
ast_cli_unregister_multiple(cli_outbound_registration, ARRAY_LEN(cli_outbound_registration));
ast_sip_unregister_cli_formatter(cli_formatter);
cli_formatter = NULL;
ast_sip_unregister_endpoint_identifier(&line_identifier);
ast_sorcery_observer_remove(ast_sip_get_sorcery(), "auth", &observer_callbacks_auth);
ast_sorcery_instance_observer_remove(ast_sip_get_sorcery(), &observer_callbacks_registrations);
ast_sorcery_object_unregister(ast_sip_get_sorcery(), "registration");
ao2_global_obj_release(current_states);
ast_sip_transport_monitor_unregister_all(registration_transport_shutdown_cb, NULL, NULL);
/* Wait for registration serializers to get destroyed. */
ast_debug(2, "Waiting for registration transactions to complete for unload.\n");
remaining = ast_serializer_shutdown_group_join(shutdown_group, MAX_UNLOAD_TIMEOUT_TIME);
if (remaining) {
/*
* NOTE: We probably have a sip_outbound_registration_client_state
* ref leak if the remaining count cannot reach zero after a few
* minutes of trying to unload.
*/
ast_log(LOG_WARNING, "Unload incomplete. Could not stop %d outbound registrations. Try again later.\n",
remaining);
return -1;
}
ast_debug(2, "Successful shutdown.\n");
ao2_cleanup(shutdown_group);
shutdown_group = NULL;
return 0;
}
static int load_module(void)
{
res_pjsip_outbound_registration: Fix reload race condition. Performing a CLI "module reload" command when there are new pjsip.conf registration objects defined frequently failed to load them correctly. What happens is a race condition between res_pjsip pushing its reload into an asynchronous task processor task and the thread that does the rest of the reloads when it gets to reloading the res_pjsip_outbound_registration module. A similar race condition happens between a reload and the CLI/AMI show registrations commands. The reload updates the current_states container and the CLI/AMI commands call get_registrations() which builds a new current_states container. * Made res_pjsip.c reload_module() use ast_sip_push_task_synchronous() instead of ast_sip_push_task() to eliminate two threads processing config reloads at the same time. * Made get_registrations() not replace the global current_states container so the CLI/AMI show registrations command cannot interfere with reloading. You could never add/remove objects in the container without the possibility of the container being replaced out from under you by get_registrations(). * Added a registration loaded sorcery instance observer to purge any dead registration objects since get_registrations() cannot do this job anymore. The struct ast_sorcery_instance_observer callbacks must be used because the callback happens inline with the load process. The struct ast_sorcery_observer callbacks are pushed to a different thread. * Added some global current_states NULL pointer checks in case the container disappears because of unload_module(). * Made sorcery's struct ast_sorcery_instance_observer.object_type_loaded callbacks guaranteed to be called before any struct ast_sorcery_observer.loaded callbacks will be called. * Moved the check for non-reloadable objects to before the sorcery instance loading callbacks happen to short circuit unnecessary work. Previously with non-reloadable objects, the sorcery instance loading/loaded callbacks would always happen, the individual wizard loading/loaded would be prevented, and the non-reloadable type logging message would be logged for each associated wizard. ASTERISK-24729 #close Review: https://reviewboard.asterisk.org/r/4381/ ........ Merged revisions 431243 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431251 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-28 04:29:23 +00:00
struct ao2_container *new_states;
shutdown_group = ast_serializer_shutdown_group_alloc();
if (!shutdown_group) {
return AST_MODULE_LOAD_DECLINE;
}
/* Create outbound registration states container. */
new_states = ao2_container_alloc_hash(AO2_ALLOC_OPT_LOCK_MUTEX, 0,
DEFAULT_STATE_BUCKETS, registration_state_hash, NULL, registration_state_cmp);
if (!new_states) {
ast_log(LOG_ERROR, "Unable to allocate registration states container\n");
unload_module();
return AST_MODULE_LOAD_DECLINE;
}
ao2_global_obj_replace_unref(current_states, new_states);
ao2_ref(new_states, -1);
/*
* Register sorcery object descriptions.
*/
ast_sorcery_apply_config(ast_sip_get_sorcery(), "res_pjsip_outbound_registration");
ast_sorcery_apply_default(ast_sip_get_sorcery(), "registration", "config", "pjsip.conf,criteria=type=registration");
if (ast_sorcery_object_register(ast_sip_get_sorcery(), "registration", sip_outbound_registration_alloc, NULL, sip_outbound_registration_apply)) {
unload_module();
return AST_MODULE_LOAD_DECLINE;
}
ast_sorcery_object_field_register(ast_sip_get_sorcery(), "registration", "type", "", OPT_NOOP_T, 0, 0);
ast_sorcery_object_field_register(ast_sip_get_sorcery(), "registration", "server_uri", "", OPT_STRINGFIELD_T, 0, STRFLDSET(struct sip_outbound_registration, server_uri));
ast_sorcery_object_field_register(ast_sip_get_sorcery(), "registration", "client_uri", "", OPT_STRINGFIELD_T, 0, STRFLDSET(struct sip_outbound_registration, client_uri));
ast_sorcery_object_field_register(ast_sip_get_sorcery(), "registration", "contact_user", "", OPT_STRINGFIELD_T, 0, STRFLDSET(struct sip_outbound_registration, contact_user));
res_pjsip: Implement additional SIP RFCs for Google Voice trunk compatability This change implements a few different generic things which were brought on by Google Voice SIP. 1. The concept of flow transports have been introduced. These are configurable transports in pjsip.conf which can be used to reference a flow of signaling to a target. These have runtime configuration that can be changed by the signaling itself (such as Service-Routes and P-Preferred-Identity). When used these guarantee an individual connection (in the case of TCP or TLS) even if multiple flow transports exist to the same target. 2. Service-Routes (RFC 3608) support has been added to the outbound registration module which when received will be stored on the flow transport and used for requests referencing it. 3. P-Associated-URI / P-Preferred-Identity (RFC 3325) support has been added to the outbound registration module. If a P-Associated-URI header is received it will be used on requests as the P-Preferred-Identity. 4. Configurable outbound extension support has been added to the outbound registration module. When set the extension will be placed in the Supported header. 5. Header parameters can now be configured on an outbound registration which will be placed in the Contact header. 6. Google specific OAuth / Bearer token authentication (draft-ietf-sipcore-sip-authn-02) has been added to the outbound registration module. All functionality changes are controlled by pjsip.conf configuration options and do not affect non-configured pjsip endpoints otherwise. ASTERISK-27971 #close Change-Id: Id214c2d1c550a41fcf564b7df8f3da7be565bd58
2018-07-18 12:45:26 +00:00
ast_sorcery_object_field_register(ast_sip_get_sorcery(), "registration", "contact_header_params", "", OPT_STRINGFIELD_T, 0, STRFLDSET(struct sip_outbound_registration, contact_header_params));
ast_sorcery_object_field_register(ast_sip_get_sorcery(), "registration", "transport", "", OPT_STRINGFIELD_T, 0, STRFLDSET(struct sip_outbound_registration, transport));
ast_sorcery_object_field_register(ast_sip_get_sorcery(), "registration", "outbound_proxy", "", OPT_STRINGFIELD_T, 0, STRFLDSET(struct sip_outbound_registration, outbound_proxy));
ast_sorcery_object_field_register(ast_sip_get_sorcery(), "registration", "expiration", "3600", OPT_UINT_T, 0, FLDSET(struct sip_outbound_registration, expiration));
ast_sorcery_object_field_register(ast_sip_get_sorcery(), "registration", "max_random_initial_delay", "10", OPT_UINT_T, 0, FLDSET(struct sip_outbound_registration, max_random_initial_delay));
ast_sorcery_object_field_register(ast_sip_get_sorcery(), "registration", "retry_interval", "60", OPT_UINT_T, 0, FLDSET(struct sip_outbound_registration, retry_interval));
ast_sorcery_object_field_register(ast_sip_get_sorcery(), "registration", "forbidden_retry_interval", "0", OPT_UINT_T, 0, FLDSET(struct sip_outbound_registration, forbidden_retry_interval));
ast_sorcery_object_field_register(ast_sip_get_sorcery(), "registration", "fatal_retry_interval", "0", OPT_UINT_T, 0, FLDSET(struct sip_outbound_registration, fatal_retry_interval));
ast_sorcery_object_field_register(ast_sip_get_sorcery(), "registration", "max_retries", "10", OPT_UINT_T, 0, FLDSET(struct sip_outbound_registration, max_retries));
ast_sorcery_object_field_register(ast_sip_get_sorcery(), "registration", "auth_rejection_permanent", "yes", OPT_BOOL_T, 1, FLDSET(struct sip_outbound_registration, auth_rejection_permanent));
sorcery: Create AST_SORCERY dialplan function. 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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410042 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-06 22:39:54 +00:00
ast_sorcery_object_field_register_custom(ast_sip_get_sorcery(), "registration", "outbound_auth", "", outbound_auth_handler, outbound_auths_to_str, outbound_auths_to_var_list, 0, 0);
ast_sorcery_object_field_register(ast_sip_get_sorcery(), "registration", "support_path", "no", OPT_BOOL_T, 1, FLDSET(struct sip_outbound_registration, support_path));
res_pjsip: Implement additional SIP RFCs for Google Voice trunk compatability This change implements a few different generic things which were brought on by Google Voice SIP. 1. The concept of flow transports have been introduced. These are configurable transports in pjsip.conf which can be used to reference a flow of signaling to a target. These have runtime configuration that can be changed by the signaling itself (such as Service-Routes and P-Preferred-Identity). When used these guarantee an individual connection (in the case of TCP or TLS) even if multiple flow transports exist to the same target. 2. Service-Routes (RFC 3608) support has been added to the outbound registration module which when received will be stored on the flow transport and used for requests referencing it. 3. P-Associated-URI / P-Preferred-Identity (RFC 3325) support has been added to the outbound registration module. If a P-Associated-URI header is received it will be used on requests as the P-Preferred-Identity. 4. Configurable outbound extension support has been added to the outbound registration module. When set the extension will be placed in the Supported header. 5. Header parameters can now be configured on an outbound registration which will be placed in the Contact header. 6. Google specific OAuth / Bearer token authentication (draft-ietf-sipcore-sip-authn-02) has been added to the outbound registration module. All functionality changes are controlled by pjsip.conf configuration options and do not affect non-configured pjsip endpoints otherwise. ASTERISK-27971 #close Change-Id: Id214c2d1c550a41fcf564b7df8f3da7be565bd58
2018-07-18 12:45:26 +00:00
ast_sorcery_object_field_register(ast_sip_get_sorcery(), "registration", "support_outbound", "no", OPT_YESNO_T, 1, FLDSET(struct sip_outbound_registration, support_outbound));
ast_sorcery_object_field_register_custom(ast_sip_get_sorcery(), "registration", "security_negotiation", "no", security_negotiation_handler, security_negotiation_to_str, NULL, 0, 0);
ast_sorcery_object_field_register_custom(ast_sip_get_sorcery(), "registration", "security_mechanisms", "", security_mechanisms_handler, security_mechanism_to_str, NULL, 0, 0);
ast_sorcery_object_field_register(ast_sip_get_sorcery(), "registration", "line", "no", OPT_BOOL_T, 1, FLDSET(struct sip_outbound_registration, line));
ast_sorcery_object_field_register(ast_sip_get_sorcery(), "registration", "endpoint", "", OPT_STRINGFIELD_T, 0, STRFLDSET(struct sip_outbound_registration, endpoint));
ast_sorcery_object_field_register(ast_sip_get_sorcery(), "registration", "user_agent", "", OPT_STRINGFIELD_T, 0, STRFLDSET(struct sip_outbound_registration, user_agent));
/*
* Register sorcery observers.
*/
if (ast_sorcery_instance_observer_add(ast_sip_get_sorcery(),
&observer_callbacks_registrations)
|| ast_sorcery_observer_add(ast_sip_get_sorcery(), "auth",
&observer_callbacks_auth)
|| ast_sorcery_observer_add(ast_sip_get_sorcery(), "registration",
&registration_observer)) {
ast_log(LOG_ERROR, "Unable to register observers.\n");
unload_module();
return AST_MODULE_LOAD_DECLINE;
}
/* Register how this module identifies endpoints. */
ast_sip_register_endpoint_identifier(&line_identifier);
/* Register CLI commands. */
cli_formatter = ao2_alloc(sizeof(struct ast_sip_cli_formatter_entry), NULL);
if (!cli_formatter) {
ast_log(LOG_ERROR, "Unable to allocate memory for cli formatter\n");
unload_module();
return AST_MODULE_LOAD_DECLINE;
}
cli_formatter->name = "registration";
cli_formatter->print_header = cli_print_header;
cli_formatter->print_body = cli_print_body;
cli_formatter->get_container = cli_get_container;
cli_formatter->iterate = cli_iterator;
cli_formatter->get_id = ast_sorcery_object_get_id;
cli_formatter->retrieve_by_id = cli_retrieve_by_id;
ast_sip_register_cli_formatter(cli_formatter);
ast_cli_register_multiple(cli_outbound_registration, ARRAY_LEN(cli_outbound_registration));
/* Register AMI actions. */
ast_manager_register_xml("PJSIPUnregister", EVENT_FLAG_SYSTEM | EVENT_FLAG_REPORTING, ami_unregister);
ast_manager_register_xml("PJSIPRegister", EVENT_FLAG_SYSTEM | EVENT_FLAG_REPORTING, ami_register);
ast_manager_register_xml("PJSIPShowRegistrationsOutbound", EVENT_FLAG_SYSTEM | EVENT_FLAG_REPORTING, ami_show_outbound_registrations);
/* Clear any previous statsd gauges in case we weren't shutdown cleanly */
ast_statsd_log("PJSIP.registrations.count", AST_STATSD_GAUGE, 0);
ast_statsd_log("PJSIP.registrations.state.Registered", AST_STATSD_GAUGE, 0);
ast_statsd_log("PJSIP.registrations.state.Unregistered", AST_STATSD_GAUGE, 0);
ast_statsd_log("PJSIP.registrations.state.Rejected", AST_STATSD_GAUGE, 0);
/* Load configuration objects */
res_pjsip_outbound_registration: Fix reload race condition. Performing a CLI "module reload" command when there are new pjsip.conf registration objects defined frequently failed to load them correctly. What happens is a race condition between res_pjsip pushing its reload into an asynchronous task processor task and the thread that does the rest of the reloads when it gets to reloading the res_pjsip_outbound_registration module. A similar race condition happens between a reload and the CLI/AMI show registrations commands. The reload updates the current_states container and the CLI/AMI commands call get_registrations() which builds a new current_states container. * Made res_pjsip.c reload_module() use ast_sip_push_task_synchronous() instead of ast_sip_push_task() to eliminate two threads processing config reloads at the same time. * Made get_registrations() not replace the global current_states container so the CLI/AMI show registrations command cannot interfere with reloading. You could never add/remove objects in the container without the possibility of the container being replaced out from under you by get_registrations(). * Added a registration loaded sorcery instance observer to purge any dead registration objects since get_registrations() cannot do this job anymore. The struct ast_sorcery_instance_observer callbacks must be used because the callback happens inline with the load process. The struct ast_sorcery_observer callbacks are pushed to a different thread. * Added some global current_states NULL pointer checks in case the container disappears because of unload_module(). * Made sorcery's struct ast_sorcery_instance_observer.object_type_loaded callbacks guaranteed to be called before any struct ast_sorcery_observer.loaded callbacks will be called. * Moved the check for non-reloadable objects to before the sorcery instance loading callbacks happen to short circuit unnecessary work. Previously with non-reloadable objects, the sorcery instance loading/loaded callbacks would always happen, the individual wizard loading/loaded would be prevented, and the non-reloadable type logging message would be logged for each associated wizard. ASTERISK-24729 #close Review: https://reviewboard.asterisk.org/r/4381/ ........ Merged revisions 431243 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431251 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-28 04:29:23 +00:00
ast_sorcery_load_object(ast_sip_get_sorcery(), "registration");
network_change_sub = stasis_subscribe(ast_system_topic(),
network_change_stasis_cb, NULL);
stasis_subscription_accept_message_type(network_change_sub, ast_network_change_type());
stasis_subscription_set_filter(network_change_sub, STASIS_SUBSCRIPTION_FILTER_SELECTIVE);
return AST_MODULE_LOAD_SUCCESS;
}
static int reload_module(void)
{
ast_sorcery_reload_object(ast_sip_get_sorcery(), "registration");
return 0;
}
AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "PJSIP Outbound Registration Support",
.support_level = AST_MODULE_SUPPORT_CORE,
.load = load_module,
.reload = reload_module,
.unload = unload_module,
.load_pri = AST_MODPRI_APP_DEPEND,
.requires = "res_pjsip",
.optional_modules = "res_statsd",
);