Merge "res_pjsip: Whitespace and comment cleanup." into 14

configs/samples/pjsip.conf.sample

@@ -672,7 +672,7 @@
                         ; usage of media encryption for this endpoint (default:
                         ; "no")
 ;media_encryption_optimistic=no ; Use encryption if possible but don't fail the call
-								; if not possible.
+                                ; if not possible.
 ;g726_non_standard=no   ; When set to "yes" and an endpoint negotiates g.726
                         ; audio then g.726 for AAL2 packing order is used contrary
                         ; to what is recommended in RFC3551. Note, 'g726aal2' also
@@ -752,7 +752,7 @@
 ;srtp_tag_32=no ; Determines whether 32 byte tags should be used instead of 80
                 ; byte tags (default: "no")
 ;set_var=       ; Variable set on a channel involving the endpoint. For multiple
-		; channel variables specify multiple 'set_var'(s)
+                ; channel variables specify multiple 'set_var'(s)
 ;rtp_keepalive= ; Interval, in seconds, between comfort noise RTP packets if
                 ; RTP is not flowing. This setting is useful for ensuring that
                 ; holes in NATs and firewalls are kept open throughout a call.
@@ -794,7 +794,7 @@
                 ; (default: "")
 ;ca_list_path=  ; Path to directory containing certificates to read TLS ONLY.
                 ; PJProject version 2.4 or higher is required for this option to
-				; be used.
+                ; be used.
                 ; (default: "")
 ;cert_file=     ; Certificate file for endpoint TLS ONLY
                 ; Will read .crt or .pem file but only uses cert,
@@ -886,8 +886,8 @@
 ;disable_tcp_switch=yes ; Disable automatic switching from UDP to TCP transports
                         ; if outgoing request is too large.
                         ; See RFC 3261 section 18.1.1.
-						; Disabling this option has been known to cause interoperability
-						; issues, so disable at your own risk.
+                        ; Disabling this option has been known to cause interoperability
+                        ; issues, so disable at your own risk.
                         ; (default: "yes")
 ;type=  ; Must be of type system (default: "")
 
@@ -917,10 +917,10 @@
 ;contact_expiration_check_interval=30
                         ; The interval (in seconds) to check for expired contacts.
 ;disable_multi_domain=no
-			; Disable Multi Domain support.
-			; If disabled it can improve realtime performace by reducing
-			; number of database requsts
-			; (default: "no")
+            ; Disable Multi Domain support.
+            ; If disabled it can improve realtime performace by reducing
+            ; number of database requsts
+            ; (default: "no")
 ;endpoint_identifier_order=ip,username,anonymous
             ; The order by which endpoint identifiers are given priority.
             ; Currently, "ip", "username", "auth_username" and "anonymous" are valid

include/asterisk/res_pjsip.h

@@ -749,9 +749,9 @@ struct ast_sip_endpoint {
 	unsigned int usereqphone;
 	/*! Whether to pass through hold and unhold using re-invites with recvonly and sendrecv */
 	unsigned int moh_passthrough;
-	/* Access control list */
+	/*! Access control list */
 	struct ast_acl_list *acl;
-	/* Restrict what IPs are allowed in the Contact header (for registration) */
+	/*! Restrict what IPs are allowed in the Contact header (for registration) */
 	struct ast_acl_list *contact_acl;
 	/*! The number of seconds into call to disable fax detection.  (0 = disabled) */
 	unsigned int faxdetect_timeout;

res/res_pjsip.c

@@ -217,10 +217,9 @@
 							<enum name="info">
 								<para>DTMF is sent as SIP INFO packets.</para>
 							</enum>
-                                                        <enum name="auto">
-                                                                <para>DTMF is sent as RFC 4733 if the other side supports it or as INBAND if not.</para>
-                                                        </enum>
-
+							<enum name="auto">
+								<para>DTMF is sent as RFC 4733 if the other side supports it or as INBAND if not.</para>
+							</enum>
 						</enumlist>
 					</description>
 				</configOption>
@@ -510,15 +509,15 @@
 				<configOption name="g726_non_standard" default="no">
 					<synopsis>Force g.726 to use AAL2 packing order when negotiating g.726 audio</synopsis>
 					<description><para>
-                                                When set to "yes" and an endpoint negotiates g.726 audio then use g.726 for AAL2
-                                                packing order instead of what is recommended by RFC3551. Since this essentially
-                                                replaces the underlying 'g726' codec with 'g726aal2' then 'g726aal2' needs to be
-                                                specified in the endpoint's allowed codec list.
+						When set to "yes" and an endpoint negotiates g.726 audio then use g.726 for AAL2
+						packing order instead of what is recommended by RFC3551. Since this essentially
+						replaces the underlying 'g726' codec with 'g726aal2' then 'g726aal2' needs to be
+						specified in the endpoint's allowed codec list.
 					</para></description>
 				</configOption>
 				<configOption name="inband_progress" default="no">
 					<synopsis>Determines whether chan_pjsip will indicate ringing using inband
-					    progress.</synopsis>
+						progress.</synopsis>
 					<description><para>
 						If set to <literal>yes</literal>, chan_pjsip will send a 183 Session Progress
 						when told to indicate ringing and will immediately start sending ringing
@@ -814,7 +813,7 @@
 				<configOption name="set_var">
 					<synopsis>Variable set on a channel involving the endpoint.</synopsis>
 					<description><para>
-					        When a new channel is created using the endpoint set the specified
+						When a new channel is created using the endpoint set the specified
 						variable(s) on that channel. For multiple channel variables specify
 						multiple 'set_var'(s).
 					</para></description>
@@ -1455,9 +1454,9 @@
 					<synopsis>Value used in User-Agent header for SIP requests and Server header for SIP responses.</synopsis>
 				</configOption>
 				<configOption name="regcontext" default="">
-                                        <synopsis>When set, Asterisk will dynamically create and destroy a NoOp priority 1 extension for a given
-					peer who registers or unregisters with us.</synopsis>
-                                </configOption>
+					<synopsis>When set, Asterisk will dynamically create and destroy a NoOp priority 1 extension for a given
+						peer who registers or unregisters with us.</synopsis>
+				</configOption>
 				<configOption name="default_outbound_endpoint" default="default_outbound_endpoint">
 					<synopsis>Endpoint to use when sending an outbound request to a URI without a specified endpoint.</synopsis>
 				</configOption>
@@ -1466,15 +1465,15 @@
 				</configOption>
 				<configOption name="debug" default="no">
 					<synopsis>Enable/Disable SIP debug logging.  Valid options include yes|no or
-                                        a host address</synopsis>
+						a host address</synopsis>
 				</configOption>
 				<configOption name="endpoint_identifier_order" default="ip,username,anonymous">
 					<synopsis>The order by which endpoint identifiers are processed and checked.
-                                        Identifier names are usually derived from and can be found in the endpoint
-                                        identifier module itself (res_pjsip_endpoint_identifier_*).
-                                        You can use the CLI command "pjsip show identifiers" to see the
-                                        identifiers currently available.</synopsis>
-                    <description>
+						Identifier names are usually derived from and can be found in the endpoint
+						identifier module itself (res_pjsip_endpoint_identifier_*).
+						You can use the CLI command "pjsip show identifiers" to see the
+						identifiers currently available.</synopsis>
+					<description>
 						<note><para>
 						One of the identifiers is "auth_username" which matches on the username in
 						an Authentication header.  This method has some security considerations because an
@@ -1488,17 +1487,17 @@
 						how many unmatched requests are received from a single ip address before a security
 						event is generated using the unidentified_request parameters.
 						</para></note>
-                    </description>
+					</description>
 				</configOption>
 				<configOption name="default_from_user" default="asterisk">
 					<synopsis>When Asterisk generates an outgoing SIP request, the From header username will be
-                                        set to this value if there is no better option (such as CallerID) to be
-                                        used.</synopsis>
+						set to this value if there is no better option (such as CallerID) to be
+						used.</synopsis>
 				</configOption>
 				<configOption name="default_realm" default="asterisk">
 					<synopsis>When Asterisk generates an challenge, the digest will be
-                                        set to this value if there is no better option (such as auth/realm) to be
-                                        used.</synopsis>
+						set to this value if there is no better option (such as auth/realm) to be
+						used.</synopsis>
 				</configOption>
 			</configObject>
 		</configFile>
@@ -2066,7 +2065,7 @@
 			Provides a listing of all endpoints.  For each endpoint an <literal>EndpointList</literal> event
 			is raised that contains relevant attributes and status information.  Once all
 			endpoints have been listed an <literal>EndpointListComplete</literal> event is issued.
-                        </para>
+			</para>
 		</description>
 		<responses>
 			<list-elements>
@@ -2102,7 +2101,7 @@
 			<literal>IdentifyDetail</literal>.  Some events may be listed multiple times if multiple objects are
 			associated (for instance AoRs).  Once all detail events have been raised a final
 			<literal>EndpointDetailComplete</literal> event is issued.
-                        </para>
+			</para>
 		</description>
 		<responses>
 			<list-elements>