Removing whitespace that causes red dots in reviewboard

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@215110 65c4cc65-6c06-0410-ace0-fbb531ad65f3

channels/chan_sip.c

@@ -103,8 +103,8 @@
  *	If someone registers a SIPS uri, this forces us to set up a TLS connection back.
  * \todo Add TCP/TLS information to function SIPPEER and SIPCHANINFO
  * \todo If tcpenable=yes, we must open a TCP socket on the same address as the IP for UDP.
- * 	 The tcpbindaddr config option should only be used to open ADDITIONAL ports
- * 	 So we should propably go back to
+ * 	The tcpbindaddr config option should only be used to open ADDITIONAL ports
+ * 	So we should propably go back to
  *		bindaddr= the default address to bind to. If tcpenable=yes, then bind this to both udp and TCP
  *				if tlsenable=yes, open TLS port (provided we also have cert)
  *		tcpbindaddr = extra address for additional TCP connections
@@ -137,7 +137,7 @@
  * \todo Default transports are set to UDP, which cause the wrong behaviour when contacting remote
  *	devices directly from the dialplan. UDP is only a fallback if no other method works,
  *	in order to be compatible with RFC2543 (SIP/1.0) devices. For transactions that exceed the
- * 	MTU (like INIVTE with video, audio and RTT)  TCP should be preferred.
+ *	MTU (like INIVTE with video, audio and RTT)  TCP should be preferred.
  *
  *	When dialling unconfigured peers (with no port number)  or devices in external domains
  *	NAPTR records MUST be consulted to find configured transport. If they are not found,
@@ -619,7 +619,7 @@ static int mwi_expiry = DEFAULT_MWI_EXPIRY;
 #define SIP_MAX_HEADERS              64               /*!< Max amount of SIP headers to read */
 #define SIP_MAX_LINES                64               /*!< Max amount of lines in SIP attachment (like SDP) */
 #define SIP_MIN_PACKET               4096             /*!< Initialize size of memory to allocate for packets */
-#define MAX_HISTORY_ENTRIES 	     50	              /*!< Max entires in the history list for a sip_pvt */
+#define MAX_HISTORY_ENTRIES		50	              /*!< Max entires in the history list for a sip_pvt */
 
 #define INITIAL_CSEQ                 101              /*!< Our initial sip sequence number */
 
@@ -641,7 +641,7 @@ static struct ast_jb_conf global_jbconf;		/*!< Global jitterbuffer configuration
 static const char config[] = "sip.conf";		/*!< Main configuration file */
 static const char notify_config[] = "sip_notify.conf";	/*!< Configuration file for sending Notify with CLI commands to reconfigure or reboot phones */
 
-#define RTP 	1
+#define RTP	1
 #define NO_RTP	0
 
 /*! \brief Authorization scheme for call transfers
@@ -730,7 +730,7 @@ static const struct cfsubscription_types {
 	const char * const text;
 } subscription_types[] = {
 	{ NONE,		   "-",        "unknown",	             "unknown" },
- 	/* RFC 4235: SIP Dialog event package */
+	/* RFC 4235: SIP Dialog event package */
 	{ DIALOG_INFO_XML, "dialog",   "application/dialog-info+xml", "dialog-info+xml" },
 	{ CPIM_PIDF_XML,   "presence", "application/cpim-pidf+xml",   "cpim-pidf+xml" },  /* RFC 3863 */
 	{ PIDF_XML,        "presence", "application/pidf+xml",        "pidf+xml" },       /* RFC 3863 */
@@ -786,13 +786,13 @@ enum sipregistrystate {
 
 	REG_STATE_REGISTERED,	/*!< Registered and done */
 
-	REG_STATE_REJECTED,	/*!< Registration rejected *
+	REG_STATE_REJECTED,	/*!< Registration rejected
 		 * \note only used when the remote party has an expire larger than
 		 * our max-expire. This is a final state from which we do not
 		 * recover (not sure how correctly).
 		 */
 
-	REG_STATE_TIMEOUT,	/*!< Registration timed out *
+	REG_STATE_TIMEOUT,	/*!< Registration timed out
 		* \note XXX unused */
 
 	REG_STATE_NOAUTH,	/*!< We have no accepted credentials
@@ -907,7 +907,7 @@ static const struct  cfsip_methods {
 	{ SIP_UNKNOWN,	 RTP,    "-UNKNOWN-", 	CAN_CREATE_DIALOG },
 	{ SIP_RESPONSE,	 NO_RTP, "SIP/2.0",	CAN_NOT_CREATE_DIALOG },
 	{ SIP_REGISTER,	 NO_RTP, "REGISTER", 	CAN_CREATE_DIALOG },
- 	{ SIP_OPTIONS,	 NO_RTP, "OPTIONS", 	CAN_CREATE_DIALOG },
+	{ SIP_OPTIONS,	 NO_RTP, "OPTIONS", 	CAN_CREATE_DIALOG },
 	{ SIP_NOTIFY,	 NO_RTP, "NOTIFY", 	CAN_CREATE_DIALOG },
 	{ SIP_INVITE,	 RTP,    "INVITE", 	CAN_CREATE_DIALOG },
 	{ SIP_ACK,	 NO_RTP, "ACK", 	CAN_NOT_CREATE_DIALOG },
@@ -924,7 +924,7 @@ static const struct  cfsip_methods {
 };
 
 /*!  Define SIP option tags, used in Require: and Supported: headers
- 	We need to be aware of these properties in the phones to use 
+	We need to be aware of these properties in the phones to use
 	the replace: header. We should not do that without knowing
 	that the other end supports it...
 	This is nothing we can configure, we learn by the dialog