[asterisk-commits] kmoore: trunk r389148 - /trunk/res/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Sun May 19 12:45:48 CDT 2013
Author: kmoore
Date: Sun May 19 12:45:42 2013
New Revision: 389148
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=389148
Log:
Add base XML documentation for res_sip
Thanks to Brad Latus, this patch adds a significant amount much-needed
documentation to res_sip. It should cover all existing configuration
options currently in Asterisk trunk.
Patch-by: Brad Latus (snuffy)
Review: https://reviewboard.asterisk.org/r/2471/
Modified:
trunk/res/res_sip.c
trunk/res/res_sip_acl.c
trunk/res/res_sip_endpoint_identifier_ip.c
trunk/res/res_sip_outbound_registration.c
Modified: trunk/res/res_sip.c
URL: http://svnview.digium.com/svn/asterisk/trunk/res/res_sip.c?view=diff&rev=389148&r1=389147&r2=389148
==============================================================================
--- trunk/res/res_sip.c (original)
+++ trunk/res/res_sip.c Sun May 19 12:45:42 2013
@@ -42,6 +42,512 @@
<support_level>core</support_level>
***/
+/*** DOCUMENTATION
+ <configInfo name="res_sip" language="en_US">
+ <synopsis>SIP Resource using PJProject</synopsis>
+ <configFile name="res_sip.conf">
+ <configObject name="endpoint">
+ <synopsis>Endpoint</synopsis>
+ <description><para>
+ The <emphasis>Endpoint</emphasis> is the primary configuration object.
+ It contains the core SIP related options only, endpoints are <emphasis>NOT</emphasis>
+ dialable entries of their own. Communication with another SIP device is
+ accomplished via Addresses of Record (AoRs) which have one or more
+ contacts assicated with them. Endpoints <emphasis>NOT</emphasis> configured to
+ use a <literal>transport</literal> will default to first transport found
+ in <filename>res_sip.conf</filename> that matches its type.
+ </para>
+ <para>Example: An Endpoint has been configured with no transport.
+ When it comes time to call an AoR, PJSIP will find the
+ first transport that matches the type. A SIP URI of <literal>sip:5000@[11::33]</literal>
+ will use the first IPv6 transport and try to send the request.
+ </para>
+ </description>
+ <configOption name="100rel" default="yes">
+ <synopsis>Allow support for RFC3262 provisional ACK tags</synopsis>
+ <description>
+ <enumlist>
+ <enum name="no" />
+ <enum name="required" />
+ <enum name="yes" />
+ </enumlist>
+ </description>
+ </configOption>
+ <configOption name="aggregate_mwi" default="yes">
+ <synopsis></synopsis>
+ <description><para>When enabled, <replaceable>aggregate_mwi</replaceable> condenses message
+ waiting notifications from multiple mailboxes into a single NOTIFY. If it is disabled,
+ individual NOTIFYs are sent for each mailbox.</para></description>
+ </configOption>
+ <configOption name="allow">
+ <synopsis>Media Codec(s) to allow</synopsis>
+ </configOption>
+ <configOption name="aors">
+ <synopsis>AoR(s) to be used with the endpoint</synopsis>
+ <description><para>
+ List of comma separated AoRs that the endpoint should be associated with.
+ </para></description>
+ </configOption>
+ <configOption name="auth">
+ <synopsis>Authentication Object(s) associated with the endpoint</synopsis>
+ <description><para>
+ This is a comma-delimited list of <replaceable>auth</replaceable> sections defined
+ in <filename>res_sip.conf</filename> to be used to verify inbound connection attempts.
+ </para><para>
+ Endpoints without an <literal>authentication</literal> object
+ configured will allow connections without vertification.
+ </para></description>
+ </configOption>
+ <configOption name="callerid">
+ <synopsis>CallerID information for the endpoint</synopsis>
+ <description><para>
+ Must be in the format <literal>Name <Number></literal>,
+ or only <literal><Number></literal>.
+ </para></description>
+ </configOption>
+ <configOption name="callerid_privacy">
+ <synopsis>Default privacy level</synopsis>
+ <description>
+ <enumlist>
+ <enum name="allowed_not_screened" />
+ <enum name="allowed_passed_screened" />
+ <enum name="allowed_failed_screened" />
+ <enum name="allowed" />
+ <enum name="prohib_not_screened" />
+ <enum name="prohib_passed_screened" />
+ <enum name="prohib_failed_screened" />
+ <enum name="prohib" />
+ <enum name="unavailable" />
+ </enumlist>
+ </description>
+ </configOption>
+ <configOption name="callerid_tag">
+ <synopsis>Internal id_tag for the endpoint</synopsis>
+ </configOption>
+ <configOption name="context">
+ <synopsis>Dialplan context for inbound sessions</synopsis>
+ </configOption>
+ <configOption name="direct_media_glare_mitigation" default="none">
+ <synopsis>Mitigation of direct media (re)INVITE glare</synopsis>
+ <description>
+ <para>
+ This setting attempts to avoid creating INVITE glare scenarios
+ by disabling direct media reINVITEs in one direction thereby allowing
+ designated servers (according to this option) to initiate direct
+ media reINVITEs without contention and significantly reducing call
+ setup time.
+ </para>
+ <para>
+ A more detailed description of how this option functions can be found on
+ the Asterisk wiki https://wiki.asterisk.org/wiki/display/AST/SIP+Direct+Media+Reinvite+Glare+Avoidance
+ </para>
+ <enumlist>
+ <enum name="none" />
+ <enum name="outgoing" />
+ <enum name="incoming" />
+ </enumlist>
+ </description>
+ </configOption>
+ <configOption name="direct_media_method" default="invite">
+ <synopsis>Direct Media method type</synopsis>
+ <description>
+ <para>Method for setting up Direct Media between endpoints.</para>
+ <enumlist>
+ <enum name="invite" />
+ <enum name="reinvite">
+ <para>Alias for the <literal>invite</literal> value.</para>
+ </enum>
+ <enum name="update" />
+ </enumlist>
+ </description>
+ </configOption>
+ <configOption name="direct_media" default="yes">
+ <synopsis>Determines whether media may flow directly between endpoints.</synopsis>
+ </configOption>
+ <configOption name="disable_direct_media_on_nat" default="no">
+ <synopsis>Disable direct media session refreshes when NAT obstructs the media session</synopsis>
+ </configOption>
+ <configOption name="disallow">
+ <synopsis>Media Codec(s) to disallow</synopsis>
+ </configOption>
+ <configOption name="dtmfmode" default="rfc4733">
+ <synopsis>DTMF mode</synopsis>
+ <description>
+ <para>This setting allows to choose the DTMF mode for endpoint communication.</para>
+ <enumlist>
+ <enum name="rfc4733">
+ <para>DTMF is sent out of band of the main audio stream.This
+ supercedes the older <emphasis>RFC-2833</emphasis> used within
+ the older <literal>chan_sip</literal>.</para>
+ </enum>
+ <enum name="inband">
+ <para>DTMF is sent as part of audio stream.</para>
+ </enum>
+ <enum name="info">
+ <para>DTMF is sent as SIP INFO packets.</para>
+ </enum>
+ </enumlist>
+ </description>
+ </configOption>
+ <configOption name="external_media_address">
+ <synopsis>IP used for External Media handling</synopsis>
+ </configOption>
+ <configOption name="force_rport" default="yes">
+ <synopsis>Force use of return port</synopsis>
+ </configOption>
+ <configOption name="ice_support" default="no">
+ <synopsis>Enable the ICE mechanism to help traverse NAT</synopsis>
+ </configOption>
+ <configOption name="identify_by" default="username,location">
+ <synopsis>Way(s) for Endpoint to be identified</synopsis>
+ <description><para>
+ There are currently two methods to identify an endpoint. By default
+ both are used to identify an endpoint.
+ </para>
+ <enumlist>
+ <enum name="username" />
+ <enum name="location" />
+ <enum name="username,location" />
+ </enumlist>
+ </description>
+ </configOption>
+ <configOption name="mailboxes">
+ <synopsis>Mailbox(es) to be associated with</synopsis>
+ </configOption>
+ <configOption name="mohsuggest" default="default">
+ <synopsis>Default Music On Hold class</synopsis>
+ </configOption>
+ <configOption name="outbound_auth">
+ <synopsis>Authentication object used for outbound requests</synopsis>
+ </configOption>
+ <configOption name="outbound_proxy">
+ <synopsis>Proxy through which to send requests</synopsis>
+ </configOption>
+ <configOption name="qualify_frequency" default="0">
+ <synopsis>Interval at which to qualify an endpoint</synopsis>
+ <description><para>
+ Interval between attempts to qualify the endpoint for reachability.
+ If <literal>0</literal> never qualify. Time in seconds.
+ </para></description>
+ </configOption>
+ <configOption name="rewrite_contact">
+ <synopsis>Allow Contact header to be rewritten with the source IP address-port</synopsis>
+ </configOption>
+ <configOption name="rtp_ipv6" default="no">
+ <synopsis>Allow use of IPv6 for RTP traffic</synopsis>
+ </configOption>
+ <configOption name="rtp_symmetric" default="no">
+ <synopsis>Enforce that RTP must be symmetric</synopsis>
+ </configOption>
+ <configOption name="send_pai" default="no">
+ <synopsis>Send the P-Asserted-Identity header</synopsis>
+ </configOption>
+ <configOption name="send_rpid" default="no">
+ <synopsis>Send the Remote-Party-ID header</synopsis>
+ </configOption>
+ <configOption name="timers_min_se" default="90">
+ <synopsis>Minimum session timers expiration period</synopsis>
+ <description><para>
+ Minimium session timer expiration period. Time in seconds.
+ </para></description>
+ </configOption>
+ <configOption name="timers" default="yes">
+ <synopsis>Session timers for SIP packets</synopsis>
+ <description>
+ <enumlist>
+ <enum name="forced" />
+ <enum name="no" />
+ <enum name="required" />
+ <enum name="yes" />
+ </enumlist>
+ </description>
+ </configOption>
+ <configOption name="timers_sess_expires" default="1800">
+ <synopsis>Maximum session timer expiration period</synopsis>
+ <description><para>
+ Maximium session timer expiration period. Time in seconds.
+ </para></description>
+ </configOption>
+ <configOption name="transport">
+ <synopsis>Desired transport configuration</synopsis>
+ <description><para>
+ This will set the desired transport configuration to send SIP data through.
+ </para>
+ <warning><para>Not specifying a transport will <emphasis>DEFAULT</emphasis>
+ to the first configured transport in <filename>res_sip.conf</filename> which is
+ valid for the URI we are trying to contact.
+ </para></warning>
+ </description>
+ </configOption>
+ <configOption name="trust_id_inbound" default="no">
+ <synopsis>Trust inbound CallerID information from endpoint</synopsis>
+ <description><para>This option determines whether res_sip will accept identification from the endpoint
+ received in a P-Asserted-Identity or Remote-Party-ID header. If <literal>no</literal>,
+ the configured Caller-ID from res_sip.conf will always be used as the identity for the
+ endpoint.</para></description>
+ </configOption>
+ <configOption name="trust_id_outbound" default="no">
+ <synopsis>Trust endpoint with private CallerID information</synopsis>
+ <description><para>This option determines whether res_sip will send identification
+ information to the endpoint that has been marked as private. If <literal>no</literal>,
+ private Caller-ID information will not be forwarded to the endpoint.</para></description>
+ </configOption>
+ <configOption name="type">
+ <synopsis>Must be of type 'endpoint'.</synopsis>
+ </configOption>
+ <configOption name="use_ptime" default="no">
+ <synopsis>Use Endpoint's requested packetisation interval</synopsis>
+ </configOption>
+ </configObject>
+ <configObject name="auth">
+ <synopsis>Authentication type</synopsis>
+ <description><para>
+ Authentication objects hold the authenitcation information for use
+ by <literal>endpoints</literal>. This also allows for multiple <literal>
+ endpoints</literal> to use the same information. Choice of MD5/plaintext
+ and setting of username.
+ </para></description>
+ <configOption name="auth_type" default="userpass">
+ <synopsis>Authentication type</synopsis>
+ <description><para>
+ This option specifies which of the password style config options should be read,
+ either 'password' or 'md5_cred' when trying to authenticate an endpoint inbound request.
+ </para>
+ <enumlist>
+ <enum name="md5"/>
+ <enum name="userpass"/>
+ </enumlist>
+ </description>
+ </configOption>
+ <configOption name="nonce_lifetime" default="32">
+ <synopsis>Lifetime of a nonce associated with this authentication config.</synopsis>
+ </configOption>
+ <configOption name="md5_cred">
+ <synopsis>MD5 Hash used for authentication.</synopsis>
+ <description><para>Only used when auth_type is <literal>md5</literal>.</para></description>
+ </configOption>
+ <configOption name="password">
+ <synopsis>PlainText password used for authentication.</synopsis>
+ <description><para>Only used when auth_type is <literal>userpass</literal>.</para></description>
+ </configOption>
+ <configOption name="realm" default="asterisk">
+ <synopsis>SIP realm for endpoint</synopsis>
+ </configOption>
+ <configOption name="type">
+ <synopsis>Must be 'auth'</synopsis>
+ </configOption>
+ <configOption name="username">
+ <synopsis>Username to use for account</synopsis>
+ </configOption>
+ </configObject>
+ <configObject name="nat_hook">
+ <synopsis>XXX This exists only to prevent XML documentation errors.</synopsis>
+ <configOption name="external_media_address">
+ <synopsis>I should be undocumented or hidden</synopsis>
+ </configOption>
+ <configOption name="method">
+ <synopsis>I should be undocumented or hidden</synopsis>
+ </configOption>
+ </configObject>
+ <configObject name="domain_alias">
+ <synopsis>Domain Alias</synopsis>
+ <description><para>
+ Signifies that a domain is an alias. Used for checking the domain of
+ the AoR to which the endpoint is binding.
+ </para></description>
+ <configOption name="type">
+ <synopsis>Must be of type 'domain_alias'.</synopsis>
+ </configOption>
+ <configOption name="domain">
+ <synopsis>Domain to be aliased</synopsis>
+ </configOption>
+ </configObject>
+ <configObject name="transport">
+ <synopsis>SIP Transport</synopsis>
+ <description><para>
+ <emphasis>Transports</emphasis>
+ </para>
+ <para>There are different transports and protocol derivatives
+ supported by <literal>res_sip</literal>. They are in order of
+ preference: UDP, TCP, and WebSocket (WS).</para>
+ <warning><para>
+ Multiple endpoints using the same connection is <emphasis>NOT</emphasis>
+ supported. Doing so may result in broken calls.
+ </para></warning>
+ </description>
+ <configOption name="async_operations" default="1">
+ <synopsis>Number of simultaneous Asynchronous Operations</synopsis>
+ </configOption>
+ <configOption name="bind">
+ <synopsis>IP Address and optional port to bind to for this transport</synopsis>
+ </configOption>
+ <configOption name="ca_list_file">
+ <synopsis>File containing a list of certificates to read (TLS ONLY)</synopsis>
+ </configOption>
+ <configOption name="cert_file">
+ <synopsis>Certificate file for endpoint (TLS ONLY)</synopsis>
+ </configOption>
+ <configOption name="cipher">
+ <synopsis>Preferred Cryptography Cipher (TLS ONLY)</synopsis>
+ <description><para>
+ Many options for acceptable ciphers see link for more:
+ http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS
+ </para></description>
+ </configOption>
+ <configOption name="domain">
+ <synopsis>Domain the transport comes from</synopsis>
+ </configOption>
+ <configOption name="external_media_address">
+ <synopsis>External Address to use in RTP handling</synopsis>
+ </configOption>
+ <configOption name="external_signaling_address">
+ <synopsis>External address for SIP signalling</synopsis>
+ </configOption>
+ <configOption name="external_signaling_port" default="0">
+ <synopsis>External port for SIP signalling</synopsis>
+ </configOption>
+ <configOption name="method">
+ <synopsis>Method of SSL transport (TLS ONLY)</synopsis>
+ <description>
+ <enumlist>
+ <enum name="default" />
+ <enum name="unspecified" />
+ <enum name="tlsv1" />
+ <enum name="sslv2" />
+ <enum name="sslv3" />
+ <enum name="sslv23" />
+ </enumlist>
+ </description>
+ </configOption>
+ <configOption name="localnet">
+ <synopsis>Network to consider local (used for NAT purposes).</synopsis>
+ <description><para>This must be in CIDR or dotted decimal format with the IP
+ and mask separated with a slash ('/').</para></description>
+ </configOption>
+ <configOption name="password">
+ <synopsis>Password required for transport</synopsis>
+ </configOption>
+ <configOption name="privkey_file">
+ <synopsis>Private key file (TLS ONLY)</synopsis>
+ </configOption>
+ <configOption name="protocol" default="udp">
+ <synopsis>Protocol to use for SIP traffic</synopsis>
+ <description>
+ <enumlist>
+ <enum name="udp" />
+ <enum name="tcp" />
+ <enum name="tls" />
+ </enumlist>
+ </description>
+ </configOption>
+ <configOption name="require_client_cert" default="false">
+ <synopsis>Require client certificate (TLS ONLY)</synopsis>
+ </configOption>
+ <configOption name="type">
+ <synopsis>Must be of type 'transport'.</synopsis>
+ </configOption>
+ <configOption name="verify_client" default="false">
+ <synopsis>Require verification of client certificate (TLS ONLY)</synopsis>
+ </configOption>
+ <configOption name="verify_server" default="false">
+ <synopsis>Require verification of server certificate (TLS ONLY)</synopsis>
+ </configOption>
+ </configObject>
+ <configObject name="contact">
+ <synopsis>A way of creating an aliased name to a SIP URI</synopsis>
+ <description><para>
+ Contacts are a way to hide SIP URIs from the dialplan directly.
+ They are also used to make a group of contactable parties when
+ in use with <literal>AoR</literal> lists.
+ </para></description>
+ <configOption name="type">
+ <synopsis>Must be of type 'contact'.</synopsis>
+ </configOption>
+ <configOption name="uri">
+ <synopsis>SIP URI to contact peer</synopsis>
+ </configOption>
+ <configOption name="expiration_time">
+ <synopsis>Time to keep alive a contact</synopsis>
+ <description><para>
+ Time to keep alive a contact. String style specification.
+ </para></description>
+ </configOption>
+ </configObject>
+ <configObject name="aor">
+ <synopsis>The configuration for a location of an endpoint</synopsis>
+ <description><para>
+ An AoR is what allows Asterisk to contact an endpoint via res_sip. If no
+ AoRs are specified, an endpoint will not be reachable by Asterisk.
+ Beyond that, an AoR has other uses within Asterisk.
+ </para><para>
+ An <literal>AoR</literal> is a way to allow dialing a group
+ of <literal>Contacts</literal> that all use the same
+ <literal>endpoint</literal> for calls.
+ </para><para>
+ This can be used as another way of grouping a list of contacts to dial
+ rather than specifing them each directly when dialing via the dialplan.
+ This must be used in conjuction with the <literal>PJSIP_DIAL_CONTACTS</literal>.
+ </para></description>
+ <configOption name="contact">
+ <synopsis>Permanent contacts assigned to AoR</synopsis>
+ <description><para>
+ Contacts included in this list will be called whenever referenced
+ by <literal>chan_pjsip</literal>.
+ </para></description>
+ </configOption>
+ <configOption name="default_expiration" default="3600">
+ <synopsis>Default expiration time in seconds for contacts that are dynamically bound to an AoR.</synopsis>
+ </configOption>
+ <configOption name="mailboxes">
+ <synopsis>Mailbox(es) to be associated with</synopsis>
+ <description><para>This option applies when an external entity subscribes to an AoR
+ for message waiting indications. The mailboxes specified here will be
+ subscribed to.</para></description>
+ </configOption>
+ <configOption name="maximum_expiration" default="7200">
+ <synopsis>Maximum time to keep an AoR</synopsis>
+ <description><para>
+ Maximium time to keep a peer with explicit expiration. Time in seconds.
+ </para></description>
+ </configOption>
+ <configOption name="max_contacts" default="0">
+ <synopsis>Maximum number of contacts that can bind to an AoR</synopsis>
+ <description><para>
+ Maximum number of contacts that can associate with this AoR.
+ </para>
+ <note><para>This should be set to <literal>1</literal> and
+ <replaceable>remove_existing</replaceable> set to <literal>yes</literal> if you
+ wish to stick with the older <literal>chan_sip</literal> behaviour.
+ </para></note>
+ </description>
+ </configOption>
+ <configOption name="minimum_expiration" default="60">
+ <synopsis>Minimum keep alive time for an AoR</synopsis>
+ <description><para>
+ Minimum time to keep a peer with an explict expiration. Time in seconds.
+ </para></description>
+ </configOption>
+ <configOption name="remove_existing" default="no">
+ <synopsis>Determines whether new contacts replace existing ones.</synopsis>
+ <description><para>
+ On receiving a new registration to the AoR should it remove
+ the existing contact that was registered against it?
+ </para>
+ <note><para>This should be set to <literal>yes</literal> and
+ <replaceable>max_contacts</replaceable> set to <literal>1</literal> if you
+ wish to stick with the older <literal>chan_sip</literal> behaviour.
+ </para></note>
+ </description>
+ </configOption>
+ <configOption name="type">
+ <synopsis>Must be of type 'aor'.</synopsis>
+ </configOption>
+ </configObject>
+ </configFile>
+ </configInfo>
+ ***/
+
+
static pjsip_endpoint *ast_pjsip_endpoint;
static struct ast_threadpool *sip_threadpool;
Modified: trunk/res/res_sip_acl.c
URL: http://svnview.digium.com/svn/asterisk/trunk/res/res_sip_acl.c?view=diff&rev=389148&r1=389147&r2=389148
==============================================================================
--- trunk/res/res_sip_acl.c (original)
+++ trunk/res/res_sip_acl.c Sun May 19 12:45:42 2013
@@ -31,6 +31,58 @@
#include "asterisk/logger.h"
#include "asterisk/sorcery.h"
#include "asterisk/acl.h"
+
+/*** DOCUMENTATION
+ <configInfo name="res_sip_acl" language="en_US">
+ <synopsis>SIP ACL module</synopsis>
+ <description><para>
+ <emphasis>ACL</emphasis>
+ </para>
+ <para>The ACL module used by <literal>res_sip</literal>. This module is
+ independent of <literal>endpoints</literal> and operates on all inbound
+ SIP communication using res_sip.
+ </para><para>
+ It should be noted that this module can also reference ACLs from
+ <filename>acl.conf</filename>.
+ </para><para>
+ There are two main ways of creating an access list: <literal>IP-Domain</literal>
+ and <literal>Contact Header</literal>. It is possible to create a combined ACL using
+ both IP and Contact.
+ </para></description>
+ <configFile name="res_sip.conf">
+ <configObject name="acl">
+ <synopsis>Access Control List</synopsis>
+ <configOption name="acl">
+ <synopsis>Name of IP ACL</synopsis>
+ <description><para>
+ This matches sections configured in <literal>acl.conf</literal>
+ </para></description>
+ </configOption>
+ <configOption name="contactacl">
+ <synopsis>Name of Contact ACL</synopsis>
+ <description><para>
+ This matches sections configured in <literal>acl.conf</literal>
+ </para></description>
+ </configOption>
+ <configOption name="contactdeny">
+ <synopsis>List of Contact Header addresses to Deny</synopsis>
+ </configOption>
+ <configOption name="contactpermit">
+ <synopsis>List of Contact Header addresses to Permit</synopsis>
+ </configOption>
+ <configOption name="deny">
+ <synopsis>List of IP-domains to deny access from</synopsis>
+ </configOption>
+ <configOption name="permit">
+ <synopsis>List of IP-domains to allow access from</synopsis>
+ </configOption>
+ <configOption name="type">
+ <synopsis>Must be of type 'acl'.</synopsis>
+ </configOption>
+ </configObject>
+ </configFile>
+ </configInfo>
+ ***/
struct sip_acl {
SORCERY_OBJECT(details);
Modified: trunk/res/res_sip_endpoint_identifier_ip.c
URL: http://svnview.digium.com/svn/asterisk/trunk/res/res_sip_endpoint_identifier_ip.c?view=diff&rev=389148&r1=389147&r2=389148
==============================================================================
--- trunk/res/res_sip_endpoint_identifier_ip.c (original)
+++ trunk/res/res_sip_endpoint_identifier_ip.c Sun May 19 12:45:42 2013
@@ -29,6 +29,25 @@
#include "asterisk/res_sip.h"
#include "asterisk/module.h"
#include "asterisk/acl.h"
+
+/*** DOCUMENTATION
+ <configInfo name="res_sip_endpoint_identifier_ip" language="en_US">
+ <synopsis>Module that identifies endpoints via source IP address</synopsis>
+ <configFile name="res_sip.conf">
+ <configObject name="identify">
+ <configOption name="endpoint">
+ <synopsis>Name of Endpoint</synopsis>
+ </configOption>
+ <configOption name="match">
+ <synopsis>IP addresses or networks to match against</synopsis>
+ </configOption>
+ <configOption name="type">
+ <synopsis>Must be of type 'identify'.</synopsis>
+ </configOption>
+ </configObject>
+ </configFile>
+ </configInfo>
+ ***/
/*! \brief Structure for an IP identification matching object */
struct ip_identify_match {
Modified: trunk/res/res_sip_outbound_registration.c
URL: http://svnview.digium.com/svn/asterisk/trunk/res/res_sip_outbound_registration.c?view=diff&rev=389148&r1=389147&r2=389148
==============================================================================
--- trunk/res/res_sip_outbound_registration.c (original)
+++ trunk/res/res_sip_outbound_registration.c Sun May 19 12:45:42 2013
@@ -30,6 +30,67 @@
#include "asterisk/res_sip.h"
#include "asterisk/module.h"
#include "asterisk/taskprocessor.h"
+
+/*** DOCUMENTATION
+ <configInfo name="res_sip_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_sip</literal> to register to other SIP servers.</para>
+ </description>
+ <configFile name="res_sip.conf">
+ <configObject name="registration">
+ <synopsis>The configuration for outbound registration</synopsis>
+ <description><para>
+ Registration is <emphasis>COMPLETELY</emphasis> separate from the rest of
+ <literal>res_sip.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>
+ </configOption>
+ <configOption name="contact_user">
+ <synopsis>Contact User to use in request</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>
+ </configOption>
+ <configOption name="outbound_auth" default="">
+ <synopsis>Authentication object to be used for outbound registrations.</synopsis>
+ </configOption>
+ <configOption name="outbound_proxy" default="">
+ <synopsis>Outbound Proxy used to send registrations</synopsis>
+ </configOption>
+ <configOption name="retry_interval" default="60">
+ <synopsis>Interval in seconds between retries if outbound registration is unsuccessful</synopsis>
+ </configOption>
+ <configOption name="server_uri">
+ <synopsis>SIP URI of the server to register against</synopsis>
+ </configOption>
+ <configOption name="transport">
+ <synopsis>Transport used for outbound authentication</synopsis>
+ <description>
+ <note><para>A <replaceable>transport</replaceable> configured in
+ <literal>res_sip.conf</literal>. As with other <literal>res_sip</literal> modules, this will use the first available transport of the appropriate type if unconfigured.</para></note>
+ </description>
+ </configOption>
+ <configOption name="type">
+ <synopsis>Must be of type 'registration'.</synopsis>
+ </configOption>
+ </configObject>
+ </configFile>
+ </configInfo>
+ ***/
/*! \brief Amount of buffer time (in seconds) before expiration that we re-register at */
#define REREGISTER_BUFFER_TIME 10
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