[asterisk-commits] mjordan: trunk r396922 - /trunk/channels/sip/include/sip.h
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Mon Aug 19 09:53:59 CDT 2013
Author: mjordan
Date: Mon Aug 19 09:53:49 2013
New Revision: 396922
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=396922
Log:
Whitespace cleanup
Remove some extraneous blobs
Modified:
trunk/channels/sip/include/sip.h
Modified: trunk/channels/sip/include/sip.h
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/sip/include/sip.h?view=diff&rev=396922&r1=396921&r2=396922
==============================================================================
--- trunk/channels/sip/include/sip.h (original)
+++ trunk/channels/sip/include/sip.h Mon Aug 19 09:53:49 2013
@@ -469,7 +469,7 @@
/*! \brief The number of media types in enum \ref media_type below. */
#define OFFERED_MEDIA_COUNT 4
-/*! \brief Media types generate different "dummy answers" for not accepting the offer of
+/*! \brief Media types generate different "dummy answers" for not accepting the offer of
a media stream. We need to add definitions for each RTP profile. Secure RTP is not
the same as normal RTP and will require a new definition */
enum media_type {
@@ -816,7 +816,7 @@
char authenticated; /*!< non-zero if this request was authenticated */
ptrdiff_t header[SIP_MAX_HEADERS]; /*!< Array of offsets into the request string of each SIP header*/
ptrdiff_t line[SIP_MAX_LINES]; /*!< Array of offsets into the request string of each SDP line*/
- struct ast_str *data;
+ struct ast_str *data;
struct ast_str *content;
/* XXX Do we need to unref socket.ser when the request goes away? */
struct sip_socket socket; /*!< The socket used for this request */
@@ -1084,7 +1084,7 @@
struct ast_format_cap *prefcaps; /*!< Preferred codec (outbound only) */
int noncodeccapability; /*!< DTMF RFC2833 telephony-event */
int jointnoncodeccapability; /*!< Joint Non codec capability */
- int maxcallbitrate; /*!< Maximum Call Bitrate for Video Calls */
+ int maxcallbitrate; /*!< Maximum Call Bitrate for Video Calls */
int t38_maxdatagram; /*!< T.38 FaxMaxDatagram override */
int request_queue_sched_id; /*!< Scheduler ID of any scheduled action to process queued requests */
int provisional_keepalive_sched_id; /*!< Scheduler ID for provisional responses that need to be sent out to avoid cancellation */
@@ -1427,7 +1427,7 @@
/*!
* \brief Definition of an MWI subscription to another server
- *
+ *
* \todo Convert this to astobj2.
*/
struct sip_subscription_mwi {
@@ -1579,7 +1579,7 @@
SIP_PUBLISH_MODIFY,
/*!
* \brief Remove
- *
+ *
* \details
* Used to remove published state from an ESC. This will contain
* an Expires header set to 0 and likely no body.
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