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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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389148 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Kinsey Moore
506
res/res_sip.c
506
res/res_sip.c
@@ -42,6 +42,512 @@
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<support_level>core</support_level>
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***/
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/*** DOCUMENTATION
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<configInfo name="res_sip" language="en_US">
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<synopsis>SIP Resource using PJProject</synopsis>
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<configFile name="res_sip.conf">
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<configObject name="endpoint">
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<synopsis>Endpoint</synopsis>
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<description><para>
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The <emphasis>Endpoint</emphasis> is the primary configuration object.
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It contains the core SIP related options only, endpoints are <emphasis>NOT</emphasis>
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dialable entries of their own. Communication with another SIP device is
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accomplished via Addresses of Record (AoRs) which have one or more
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contacts assicated with them. Endpoints <emphasis>NOT</emphasis> configured to
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use a <literal>transport</literal> will default to first transport found
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in <filename>res_sip.conf</filename> that matches its type.
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</para>
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<para>Example: An Endpoint has been configured with no transport.
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When it comes time to call an AoR, PJSIP will find the
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first transport that matches the type. A SIP URI of <literal>sip:5000@[11::33]</literal>
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will use the first IPv6 transport and try to send the request.
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</para>
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</description>
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<configOption name="100rel" default="yes">
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<synopsis>Allow support for RFC3262 provisional ACK tags</synopsis>
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<description>
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<enumlist>
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<enum name="no" />
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<enum name="required" />
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<enum name="yes" />
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</enumlist>
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</description>
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</configOption>
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<configOption name="aggregate_mwi" default="yes">
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<synopsis></synopsis>
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<description><para>When enabled, <replaceable>aggregate_mwi</replaceable> condenses message
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waiting notifications from multiple mailboxes into a single NOTIFY. If it is disabled,
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individual NOTIFYs are sent for each mailbox.</para></description>
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</configOption>
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<configOption name="allow">
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<synopsis>Media Codec(s) to allow</synopsis>
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</configOption>
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<configOption name="aors">
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<synopsis>AoR(s) to be used with the endpoint</synopsis>
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<description><para>
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List of comma separated AoRs that the endpoint should be associated with.
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</para></description>
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</configOption>
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<configOption name="auth">
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<synopsis>Authentication Object(s) associated with the endpoint</synopsis>
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<description><para>
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This is a comma-delimited list of <replaceable>auth</replaceable> sections defined
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in <filename>res_sip.conf</filename> to be used to verify inbound connection attempts.
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</para><para>
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Endpoints without an <literal>authentication</literal> object
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configured will allow connections without vertification.
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</para></description>
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</configOption>
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<configOption name="callerid">
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<synopsis>CallerID information for the endpoint</synopsis>
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<description><para>
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Must be in the format <literal>Name <Number></literal>,
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or only <literal><Number></literal>.
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</para></description>
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</configOption>
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<configOption name="callerid_privacy">
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<synopsis>Default privacy level</synopsis>
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<description>
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<enumlist>
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<enum name="allowed_not_screened" />
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<enum name="allowed_passed_screened" />
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<enum name="allowed_failed_screened" />
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<enum name="allowed" />
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<enum name="prohib_not_screened" />
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<enum name="prohib_passed_screened" />
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<enum name="prohib_failed_screened" />
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<enum name="prohib" />
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<enum name="unavailable" />
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</enumlist>
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</description>
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</configOption>
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<configOption name="callerid_tag">
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<synopsis>Internal id_tag for the endpoint</synopsis>
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</configOption>
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<configOption name="context">
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<synopsis>Dialplan context for inbound sessions</synopsis>
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</configOption>
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<configOption name="direct_media_glare_mitigation" default="none">
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<synopsis>Mitigation of direct media (re)INVITE glare</synopsis>
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<description>
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<para>
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This setting attempts to avoid creating INVITE glare scenarios
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by disabling direct media reINVITEs in one direction thereby allowing
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designated servers (according to this option) to initiate direct
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media reINVITEs without contention and significantly reducing call
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setup time.
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</para>
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<para>
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A more detailed description of how this option functions can be found on
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the Asterisk wiki https://wiki.asterisk.org/wiki/display/AST/SIP+Direct+Media+Reinvite+Glare+Avoidance
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</para>
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<enumlist>
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<enum name="none" />
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<enum name="outgoing" />
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<enum name="incoming" />
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</enumlist>
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</description>
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</configOption>
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<configOption name="direct_media_method" default="invite">
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<synopsis>Direct Media method type</synopsis>
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<description>
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<para>Method for setting up Direct Media between endpoints.</para>
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<enumlist>
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<enum name="invite" />
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<enum name="reinvite">
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<para>Alias for the <literal>invite</literal> value.</para>
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</enum>
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<enum name="update" />
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</enumlist>
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</description>
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</configOption>
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<configOption name="direct_media" default="yes">
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<synopsis>Determines whether media may flow directly between endpoints.</synopsis>
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</configOption>
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<configOption name="disable_direct_media_on_nat" default="no">
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<synopsis>Disable direct media session refreshes when NAT obstructs the media session</synopsis>
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</configOption>
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<configOption name="disallow">
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<synopsis>Media Codec(s) to disallow</synopsis>
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</configOption>
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<configOption name="dtmfmode" default="rfc4733">
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<synopsis>DTMF mode</synopsis>
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<description>
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<para>This setting allows to choose the DTMF mode for endpoint communication.</para>
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<enumlist>
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<enum name="rfc4733">
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<para>DTMF is sent out of band of the main audio stream.This
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supercedes the older <emphasis>RFC-2833</emphasis> used within
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the older <literal>chan_sip</literal>.</para>
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</enum>
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<enum name="inband">
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<para>DTMF is sent as part of audio stream.</para>
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</enum>
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<enum name="info">
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<para>DTMF is sent as SIP INFO packets.</para>
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</enum>
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</enumlist>
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</description>
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</configOption>
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<configOption name="external_media_address">
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<synopsis>IP used for External Media handling</synopsis>
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</configOption>
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<configOption name="force_rport" default="yes">
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<synopsis>Force use of return port</synopsis>
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</configOption>
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<configOption name="ice_support" default="no">
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<synopsis>Enable the ICE mechanism to help traverse NAT</synopsis>
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</configOption>
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<configOption name="identify_by" default="username,location">
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<synopsis>Way(s) for Endpoint to be identified</synopsis>
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<description><para>
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There are currently two methods to identify an endpoint. By default
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both are used to identify an endpoint.
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</para>
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<enumlist>
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<enum name="username" />
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<enum name="location" />
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<enum name="username,location" />
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</enumlist>
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</description>
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</configOption>
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<configOption name="mailboxes">
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<synopsis>Mailbox(es) to be associated with</synopsis>
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</configOption>
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<configOption name="mohsuggest" default="default">
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<synopsis>Default Music On Hold class</synopsis>
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</configOption>
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<configOption name="outbound_auth">
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<synopsis>Authentication object used for outbound requests</synopsis>
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</configOption>
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<configOption name="outbound_proxy">
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<synopsis>Proxy through which to send requests</synopsis>
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</configOption>
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<configOption name="qualify_frequency" default="0">
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<synopsis>Interval at which to qualify an endpoint</synopsis>
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<description><para>
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Interval between attempts to qualify the endpoint for reachability.
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If <literal>0</literal> never qualify. Time in seconds.
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</para></description>
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</configOption>
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<configOption name="rewrite_contact">
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<synopsis>Allow Contact header to be rewritten with the source IP address-port</synopsis>
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</configOption>
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<configOption name="rtp_ipv6" default="no">
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<synopsis>Allow use of IPv6 for RTP traffic</synopsis>
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</configOption>
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<configOption name="rtp_symmetric" default="no">
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<synopsis>Enforce that RTP must be symmetric</synopsis>
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</configOption>
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<configOption name="send_pai" default="no">
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<synopsis>Send the P-Asserted-Identity header</synopsis>
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</configOption>
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<configOption name="send_rpid" default="no">
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<synopsis>Send the Remote-Party-ID header</synopsis>
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</configOption>
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<configOption name="timers_min_se" default="90">
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<synopsis>Minimum session timers expiration period</synopsis>
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<description><para>
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Minimium session timer expiration period. Time in seconds.
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</para></description>
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</configOption>
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<configOption name="timers" default="yes">
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<synopsis>Session timers for SIP packets</synopsis>
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<description>
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<enumlist>
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<enum name="forced" />
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<enum name="no" />
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<enum name="required" />
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<enum name="yes" />
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</enumlist>
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</description>
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</configOption>
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<configOption name="timers_sess_expires" default="1800">
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<synopsis>Maximum session timer expiration period</synopsis>
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<description><para>
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Maximium session timer expiration period. Time in seconds.
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</para></description>
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</configOption>
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<configOption name="transport">
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<synopsis>Desired transport configuration</synopsis>
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<description><para>
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This will set the desired transport configuration to send SIP data through.
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</para>
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<warning><para>Not specifying a transport will <emphasis>DEFAULT</emphasis>
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to the first configured transport in <filename>res_sip.conf</filename> which is
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valid for the URI we are trying to contact.
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</para></warning>
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</description>
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</configOption>
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<configOption name="trust_id_inbound" default="no">
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<synopsis>Trust inbound CallerID information from endpoint</synopsis>
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<description><para>This option determines whether res_sip will accept identification from the endpoint
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received in a P-Asserted-Identity or Remote-Party-ID header. If <literal>no</literal>,
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the configured Caller-ID from res_sip.conf will always be used as the identity for the
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endpoint.</para></description>
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</configOption>
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<configOption name="trust_id_outbound" default="no">
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<synopsis>Trust endpoint with private CallerID information</synopsis>
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<description><para>This option determines whether res_sip will send identification
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information to the endpoint that has been marked as private. If <literal>no</literal>,
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private Caller-ID information will not be forwarded to the endpoint.</para></description>
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</configOption>
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<configOption name="type">
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<synopsis>Must be of type 'endpoint'.</synopsis>
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</configOption>
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<configOption name="use_ptime" default="no">
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<synopsis>Use Endpoint's requested packetisation interval</synopsis>
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</configOption>
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</configObject>
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<configObject name="auth">
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<synopsis>Authentication type</synopsis>
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<description><para>
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Authentication objects hold the authenitcation information for use
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by <literal>endpoints</literal>. This also allows for multiple <literal>
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endpoints</literal> to use the same information. Choice of MD5/plaintext
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and setting of username.
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</para></description>
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<configOption name="auth_type" default="userpass">
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<synopsis>Authentication type</synopsis>
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<description><para>
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This option specifies which of the password style config options should be read,
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either 'password' or 'md5_cred' when trying to authenticate an endpoint inbound request.
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</para>
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<enumlist>
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<enum name="md5"/>
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<enum name="userpass"/>
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</enumlist>
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</description>
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</configOption>
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<configOption name="nonce_lifetime" default="32">
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<synopsis>Lifetime of a nonce associated with this authentication config.</synopsis>
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</configOption>
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<configOption name="md5_cred">
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<synopsis>MD5 Hash used for authentication.</synopsis>
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<description><para>Only used when auth_type is <literal>md5</literal>.</para></description>
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</configOption>
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<configOption name="password">
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<synopsis>PlainText password used for authentication.</synopsis>
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<description><para>Only used when auth_type is <literal>userpass</literal>.</para></description>
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</configOption>
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<configOption name="realm" default="asterisk">
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<synopsis>SIP realm for endpoint</synopsis>
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</configOption>
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<configOption name="type">
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<synopsis>Must be 'auth'</synopsis>
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</configOption>
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<configOption name="username">
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<synopsis>Username to use for account</synopsis>
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</configOption>
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</configObject>
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<configObject name="nat_hook">
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<synopsis>XXX This exists only to prevent XML documentation errors.</synopsis>
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<configOption name="external_media_address">
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<synopsis>I should be undocumented or hidden</synopsis>
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</configOption>
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<configOption name="method">
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<synopsis>I should be undocumented or hidden</synopsis>
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</configOption>
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</configObject>
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<configObject name="domain_alias">
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<synopsis>Domain Alias</synopsis>
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<description><para>
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Signifies that a domain is an alias. Used for checking the domain of
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the AoR to which the endpoint is binding.
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</para></description>
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<configOption name="type">
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<synopsis>Must be of type 'domain_alias'.</synopsis>
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</configOption>
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<configOption name="domain">
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<synopsis>Domain to be aliased</synopsis>
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</configOption>
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</configObject>
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<configObject name="transport">
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<synopsis>SIP Transport</synopsis>
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<description><para>
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<emphasis>Transports</emphasis>
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</para>
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<para>There are different transports and protocol derivatives
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supported by <literal>res_sip</literal>. They are in order of
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preference: UDP, TCP, and WebSocket (WS).</para>
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<warning><para>
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Multiple endpoints using the same connection is <emphasis>NOT</emphasis>
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supported. Doing so may result in broken calls.
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</para></warning>
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</description>
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<configOption name="async_operations" default="1">
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<synopsis>Number of simultaneous Asynchronous Operations</synopsis>
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</configOption>
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<configOption name="bind">
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<synopsis>IP Address and optional port to bind to for this transport</synopsis>
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</configOption>
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<configOption name="ca_list_file">
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<synopsis>File containing a list of certificates to read (TLS ONLY)</synopsis>
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</configOption>
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<configOption name="cert_file">
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<synopsis>Certificate file for endpoint (TLS ONLY)</synopsis>
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</configOption>
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<configOption name="cipher">
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<synopsis>Preferred Cryptography Cipher (TLS ONLY)</synopsis>
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<description><para>
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Many options for acceptable ciphers see link for more:
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http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS
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</para></description>
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</configOption>
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<configOption name="domain">
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<synopsis>Domain the transport comes from</synopsis>
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</configOption>
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<configOption name="external_media_address">
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<synopsis>External Address to use in RTP handling</synopsis>
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</configOption>
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<configOption name="external_signaling_address">
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<synopsis>External address for SIP signalling</synopsis>
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</configOption>
|
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<configOption name="external_signaling_port" default="0">
|
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<synopsis>External port for SIP signalling</synopsis>
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||||
</configOption>
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||||
<configOption name="method">
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||||
<synopsis>Method of SSL transport (TLS ONLY)</synopsis>
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||||
<description>
|
||||
<enumlist>
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||||
<enum name="default" />
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||||
<enum name="unspecified" />
|
||||
<enum name="tlsv1" />
|
||||
<enum name="sslv2" />
|
||||
<enum name="sslv3" />
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||||
<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;
|
||||
|
||||
@@ -32,6 +32,58 @@
|
||||
#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);
|
||||
struct ast_acl_list *acl;
|
||||
|
||||
@@ -30,6 +30,25 @@
|
||||
#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 {
|
||||
/*! \brief Sorcery object details */
|
||||
|
||||
@@ -31,6 +31,67 @@
|
||||
#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
|
||||
|
||||
|
||||
Reference in New Issue
Block a user