Merge "res_pjsip: Whitespace and comment cleanup."
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zuul
Gerrit Code Review
commit
8e79e382b4
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@ -672,7 +672,7 @@
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; usage of media encryption for this endpoint (default:
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; "no")
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;media_encryption_optimistic=no ; Use encryption if possible but don't fail the call
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; if not possible.
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; if not possible.
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;g726_non_standard=no ; When set to "yes" and an endpoint negotiates g.726
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; audio then g.726 for AAL2 packing order is used contrary
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; to what is recommended in RFC3551. Note, 'g726aal2' also
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@ -752,7 +752,7 @@
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;srtp_tag_32=no ; Determines whether 32 byte tags should be used instead of 80
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; byte tags (default: "no")
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;set_var= ; Variable set on a channel involving the endpoint. For multiple
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; channel variables specify multiple 'set_var'(s)
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; channel variables specify multiple 'set_var'(s)
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;rtp_keepalive= ; Interval, in seconds, between comfort noise RTP packets if
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; RTP is not flowing. This setting is useful for ensuring that
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; holes in NATs and firewalls are kept open throughout a call.
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@ -794,7 +794,7 @@
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; (default: "")
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;ca_list_path= ; Path to directory containing certificates to read TLS ONLY.
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; PJProject version 2.4 or higher is required for this option to
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; be used.
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; be used.
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; (default: "")
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;cert_file= ; Certificate file for endpoint TLS ONLY
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; Will read .crt or .pem file but only uses cert,
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@ -886,8 +886,8 @@
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;disable_tcp_switch=yes ; Disable automatic switching from UDP to TCP transports
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; if outgoing request is too large.
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; See RFC 3261 section 18.1.1.
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; Disabling this option has been known to cause interoperability
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; issues, so disable at your own risk.
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; Disabling this option has been known to cause interoperability
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; issues, so disable at your own risk.
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; (default: "yes")
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;type= ; Must be of type system (default: "")
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@ -917,10 +917,10 @@
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;contact_expiration_check_interval=30
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; The interval (in seconds) to check for expired contacts.
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;disable_multi_domain=no
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; Disable Multi Domain support.
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; If disabled it can improve realtime performace by reducing
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; number of database requsts
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; (default: "no")
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; Disable Multi Domain support.
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; If disabled it can improve realtime performace by reducing
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; number of database requsts
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; (default: "no")
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;endpoint_identifier_order=ip,username,anonymous
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; The order by which endpoint identifiers are given priority.
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; Currently, "ip", "username", "auth_username" and "anonymous" are valid
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@ -749,9 +749,9 @@ struct ast_sip_endpoint {
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unsigned int usereqphone;
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/*! Whether to pass through hold and unhold using re-invites with recvonly and sendrecv */
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unsigned int moh_passthrough;
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/* Access control list */
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/*! Access control list */
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struct ast_acl_list *acl;
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/* Restrict what IPs are allowed in the Contact header (for registration) */
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/*! Restrict what IPs are allowed in the Contact header (for registration) */
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struct ast_acl_list *contact_acl;
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/*! The number of seconds into call to disable fax detection. (0 = disabled) */
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unsigned int faxdetect_timeout;
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@ -217,10 +217,9 @@
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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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<enum name="auto">
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<para>DTMF is sent as RFC 4733 if the other side supports it or as INBAND if not.</para>
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</enum>
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<enum name="auto">
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<para>DTMF is sent as RFC 4733 if the other side supports it or as INBAND if not.</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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@ -510,15 +509,15 @@
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<configOption name="g726_non_standard" default="no">
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<synopsis>Force g.726 to use AAL2 packing order when negotiating g.726 audio</synopsis>
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<description><para>
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When set to "yes" and an endpoint negotiates g.726 audio then use g.726 for AAL2
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packing order instead of what is recommended by RFC3551. Since this essentially
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replaces the underlying 'g726' codec with 'g726aal2' then 'g726aal2' needs to be
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specified in the endpoint's allowed codec list.
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When set to "yes" and an endpoint negotiates g.726 audio then use g.726 for AAL2
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packing order instead of what is recommended by RFC3551. Since this essentially
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replaces the underlying 'g726' codec with 'g726aal2' then 'g726aal2' needs to be
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specified in the endpoint's allowed codec list.
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</para></description>
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</configOption>
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<configOption name="inband_progress" default="no">
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<synopsis>Determines whether chan_pjsip will indicate ringing using inband
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progress.</synopsis>
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progress.</synopsis>
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<description><para>
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If set to <literal>yes</literal>, chan_pjsip will send a 183 Session Progress
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when told to indicate ringing and will immediately start sending ringing
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@ -814,7 +813,7 @@
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<configOption name="set_var">
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<synopsis>Variable set on a channel involving the endpoint.</synopsis>
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<description><para>
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When a new channel is created using the endpoint set the specified
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When a new channel is created using the endpoint set the specified
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variable(s) on that channel. For multiple channel variables specify
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multiple 'set_var'(s).
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</para></description>
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@ -1455,9 +1454,9 @@
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<synopsis>Value used in User-Agent header for SIP requests and Server header for SIP responses.</synopsis>
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</configOption>
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<configOption name="regcontext" default="">
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<synopsis>When set, Asterisk will dynamically create and destroy a NoOp priority 1 extension for a given
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peer who registers or unregisters with us.</synopsis>
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</configOption>
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<synopsis>When set, Asterisk will dynamically create and destroy a NoOp priority 1 extension for a given
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peer who registers or unregisters with us.</synopsis>
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</configOption>
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<configOption name="default_outbound_endpoint" default="default_outbound_endpoint">
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<synopsis>Endpoint to use when sending an outbound request to a URI without a specified endpoint.</synopsis>
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</configOption>
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@ -1466,15 +1465,15 @@
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</configOption>
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<configOption name="debug" default="no">
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<synopsis>Enable/Disable SIP debug logging. Valid options include yes|no or
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a host address</synopsis>
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a host address</synopsis>
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</configOption>
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<configOption name="endpoint_identifier_order" default="ip,username,anonymous">
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<synopsis>The order by which endpoint identifiers are processed and checked.
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Identifier names are usually derived from and can be found in the endpoint
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identifier module itself (res_pjsip_endpoint_identifier_*).
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You can use the CLI command "pjsip show identifiers" to see the
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identifiers currently available.</synopsis>
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<description>
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Identifier names are usually derived from and can be found in the endpoint
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identifier module itself (res_pjsip_endpoint_identifier_*).
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You can use the CLI command "pjsip show identifiers" to see the
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identifiers currently available.</synopsis>
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<description>
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<note><para>
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One of the identifiers is "auth_username" which matches on the username in
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an Authentication header. This method has some security considerations because an
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@ -1488,17 +1487,17 @@
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how many unmatched requests are received from a single ip address before a security
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event is generated using the unidentified_request parameters.
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</para></note>
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</description>
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</description>
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</configOption>
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<configOption name="default_from_user" default="asterisk">
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<synopsis>When Asterisk generates an outgoing SIP request, the From header username will be
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set to this value if there is no better option (such as CallerID) to be
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used.</synopsis>
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set to this value if there is no better option (such as CallerID) to be
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used.</synopsis>
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</configOption>
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<configOption name="default_realm" default="asterisk">
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<synopsis>When Asterisk generates an challenge, the digest will be
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set to this value if there is no better option (such as auth/realm) to be
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used.</synopsis>
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set to this value if there is no better option (such as auth/realm) to be
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used.</synopsis>
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</configOption>
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</configObject>
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</configFile>
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@ -2066,7 +2065,7 @@
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Provides a listing of all endpoints. For each endpoint an <literal>EndpointList</literal> event
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is raised that contains relevant attributes and status information. Once all
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endpoints have been listed an <literal>EndpointListComplete</literal> event is issued.
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</para>
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</para>
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</description>
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<responses>
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<list-elements>
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@ -2102,7 +2101,7 @@
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<literal>IdentifyDetail</literal>. Some events may be listed multiple times if multiple objects are
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associated (for instance AoRs). Once all detail events have been raised a final
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<literal>EndpointDetailComplete</literal> event is issued.
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</para>
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</para>
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</description>
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<responses>
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<list-elements>
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