[svn-commits] russell: trunk r48417 - in /trunk: channels/
include/asterisk/
svn-commits at lists.digium.com
svn-commits at lists.digium.com
Tue Dec 12 15:32:20 MST 2006
Author: russell
Date: Tue Dec 12 16:32:20 2006
New Revision: 48417
URL: http://svn.digium.com/view/asterisk?view=rev&rev=48417
Log:
Fix various spelling mistakes in comments.
Modified:
trunk/channels/chan_sip.c
trunk/include/asterisk/app.h
trunk/include/asterisk/channel.h
trunk/include/asterisk/pbx.h
Modified: trunk/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/trunk/channels/chan_sip.c?view=diff&rev=48417&r1=48416&r2=48417
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Tue Dec 12 16:32:20 2006
@@ -811,7 +811,7 @@
#define T38FAX_RATE_12000 (1 << 12) /*!< 12000 bps t38FaxRate */
#define T38FAX_RATE_14400 (1 << 13) /*!< 14400 bps t38FaxRate */
-/*!< This is default: NO MMR and JBIG trancoding, NO fill bit removal, transferredTCF TCF, UDP FEC, Version 0 and 9600 max fax rate */
+/*!< This is default: NO MMR and JBIG transcoding, NO fill bit removal, transferredTCF TCF, UDP FEC, Version 0 and 9600 max fax rate */
static int global_t38_capability = T38FAX_VERSION_0 | T38FAX_RATE_2400 | T38FAX_RATE_4800 | T38FAX_RATE_7200 | T38FAX_RATE_9600;
#define sipdebug ast_test_flag(&global_flags[1], SIP_PAGE2_DEBUG)
@@ -841,7 +841,7 @@
enum referstatus {
REFER_IDLE, /*!< No REFER is in progress */
REFER_SENT, /*!< Sent REFER to transferee */
- REFER_RECEIVED, /*!< Received REFER from transferer */
+ REFER_RECEIVED, /*!< Received REFER from transferrer */
REFER_CONFIRMED, /*!< Refer confirmed with a 100 TRYING */
REFER_ACCEPTED, /*!< Accepted by transferee */
REFER_RINGING, /*!< Target Ringing */
@@ -1159,7 +1159,7 @@
int refresh; /*!< How often to refresh */
struct sip_pvt *call; /*!< create a sip_pvt structure for each outbound "registration dialog" in progress */
enum sipregistrystate regstate; /*!< Registration state (see above) */
- time_t regtime; /*!< Last succesful registration time */
+ time_t regtime; /*!< Last successful registration time */
int callid_valid; /*!< 0 means we haven't chosen callid for this registry yet. */
unsigned int ocseq; /*!< Sequence number we got to for REGISTERs for this registry */
struct sockaddr_in us; /*!< Who the server thinks we are */
@@ -1179,7 +1179,7 @@
ASTOBJ_CONTAINER_COMPONENTS(struct sip_peer);
} peerl;
-/*! \brief The register list: Other SIP proxys we register with and place calls to */
+/*! \brief The register list: Other SIP proxies we register with and place calls to */
static struct ast_register_list {
ASTOBJ_CONTAINER_COMPONENTS(struct sip_registry);
int recheck;
@@ -2872,7 +2872,7 @@
struct sip_pvt *p;
struct varshead *headp;
struct ast_var_t *current;
- const char *referer = NULL; /* SIP refererer */
+ const char *referer = NULL; /* SIP referrer */
p = ast->tech_pvt;
if ((ast->_state != AST_STATE_DOWN) && (ast->_state != AST_STATE_RESERVED)) {
@@ -2895,7 +2895,7 @@
/* This is a transfered call */
p->options->transfer = 1;
} else if (!strcasecmp(ast_var_name(current), "SIPTRANSFER_REFERER")) {
- /* This is the referer */
+ /* This is the referrer */
referer = ast_var_value(current);
} else if (!strcasecmp(ast_var_name(current), "SIPTRANSFER_REPLACES")) {
/* We're replacing a call. */
@@ -3082,9 +3082,9 @@
* This will cause unexpected behaviour in subscriptions, since a "friend"
* is *two* devices in Asterisk, not one.
*
- * Thought: For realtime, we should propably update storage with inuse counter...
+ * Thought: For realtime, we should probably update storage with inuse counter...
*
- * \return 0 if call is ok (no call limit, below treshold)
+ * \return 0 if call is ok (no call limit, below threshold)
* -1 on rejection of call
*
*/
@@ -3244,7 +3244,7 @@
return AST_CAUSE_NO_ANSWER;
case 484: /* Address incomplete */
return AST_CAUSE_INVALID_NUMBER_FORMAT;
- case 485: /* Ambigous */
+ case 485: /* Ambiguous */
return AST_CAUSE_UNALLOCATED;
case 486: /* Busy everywhere */
return AST_CAUSE_BUSY;
Modified: trunk/include/asterisk/app.h
URL: http://svn.digium.com/view/asterisk/trunk/include/asterisk/app.h?view=diff&rev=48417&r1=48416&r2=48417
==============================================================================
--- trunk/include/asterisk/app.h (original)
+++ trunk/include/asterisk/app.h Tue Dec 12 16:32:20 2006
@@ -127,7 +127,7 @@
* Normally, Asterisk has a SIGCHLD handler that is cleaning up all zombie
* processes from forking elsewhere in Asterisk. However, if you want to
* wait*() on the process to retrieve information about it's exit status,
- * then this signal handler needs to be temporaraly replaced.
+ * then this signal handler needs to be temporarily replaced.
*
* Code that executes this function *must* call ast_unreplace_sigchld()
* after it is finished doing the wait*().
Modified: trunk/include/asterisk/channel.h
URL: http://svn.digium.com/view/asterisk/trunk/include/asterisk/channel.h?view=diff&rev=48417&r1=48416&r2=48417
==============================================================================
--- trunk/include/asterisk/channel.h (original)
+++ trunk/include/asterisk/channel.h Tue Dec 12 16:32:20 2006
@@ -91,7 +91,7 @@
The Core bridge handles two channels (a "phone call") and bridge
them together.
- The conference bridge (meetme) handles several channels simultaneosly
+ The conference bridge (meetme) handles several channels simultaneously
with the support of an external timer (zaptel timer). This is used
not only by the Conference application (meetme) but also by the
page application and the SLA system introduced in 1.4.
@@ -679,7 +679,7 @@
* \param format requested channel format
* \param data data to pass to the channel requester
* \param timeout maximum amount of time to wait for an answer
- * \param reason why unsuccessful (if unsuceessful)
+ * \param reason why unsuccessful (if unsuccessful)
* \param cidnum Caller-ID Number
* \param cidname Caller-ID Name
* \return Returns an ast_channel on success or no answer, NULL on failure. Check the value of chan->_state
@@ -745,7 +745,7 @@
* This function compares a offset from current time with the absolute time
* out on a channel (when to hang up). If the absolute time out on a channel
* is earlier than current time plus the offset, it returns 1, if the two
- * time values are equal, it return 0, otherwise, it retturn -1.
+ * time values are equal, it return 0, otherwise, it return -1.
*/
int ast_channel_cmpwhentohangup(struct ast_channel *chan, time_t offset);
@@ -802,7 +802,7 @@
\return Returns < 0 on failure, 0 if nothing ever arrived, and the # of ms remaining otherwise */
int ast_waitfor(struct ast_channel *chan, int ms);
-/*! \brief Wait for a specied amount of time, looking for hangups
+/*! \brief Wait for a specified amount of time, looking for hangups
* \param chan channel to wait for
* \param ms length of time in milliseconds to sleep
* Waits for a specified amount of time, servicing the channel as required.
@@ -810,7 +810,7 @@
*/
int ast_safe_sleep(struct ast_channel *chan, int ms);
-/*! \brief Wait for a specied amount of time, looking for hangups and a condition argument
+/*! \brief Wait for a specified amount of time, looking for hangups and a condition argument
* \param chan channel to wait for
* \param ms length of time in milliseconds to sleep
* \param cond a function pointer for testing continue condition
@@ -893,7 +893,7 @@
int ast_set_read_format(struct ast_channel *chan, int format);
/*! \brief Sets write format on channel chan
- * Set write format for channel to whichever compoent of "format" is best.
+ * Set write format for channel to whichever component of "format" is best.
* \param chan channel to change
* \param format new format for writing
* \return Returns 0 on success, -1 on failure
@@ -1078,7 +1078,7 @@
* \param transfercapability transfercapabilty to get the name of
* Give a name to a transfercapbility
* See above
- * Returns the text form of the binary transfer capbility
+ * Returns the text form of the binary transfer capbaility
*/
char *ast_transfercapability2str(int transfercapability) attribute_const;
@@ -1129,7 +1129,6 @@
being DTMF-deferred */
int ast_channel_defer_dtmf(struct ast_channel *chan);
-/*! Undeos a defer */
/*! Undo defer. ast_read will return any dtmf characters that were queued */
void ast_channel_undefer_dtmf(struct ast_channel *chan);
@@ -1150,7 +1149,7 @@
/*! Activate a given generator */
int ast_activate_generator(struct ast_channel *chan, struct ast_generator *gen, void *params);
-/*! Deactive an active generator */
+/*! Deactivate an active generator */
void ast_deactivate_generator(struct ast_channel *chan);
void ast_set_callerid(struct ast_channel *chan, const char *cidnum, const char *cidname, const char *ani);
@@ -1196,7 +1195,7 @@
int ast_transfer(struct ast_channel *chan, char *dest);
/*! \brief Start masquerading a channel
- XXX This is a seriously wacked out operation. We're essentially putting the guts of
+ XXX This is a seriously whacked out operation. We're essentially putting the guts of
the clone channel into the original channel. Start by killing off the original
channel's backend. I'm not sure we're going to keep this function, because
while the features are nice, the cost is very high in terms of pure nastiness. XXX
Modified: trunk/include/asterisk/pbx.h
URL: http://svn.digium.com/view/asterisk/trunk/include/asterisk/pbx.h?view=diff&rev=48417&r1=48416&r2=48417
==============================================================================
--- trunk/include/asterisk/pbx.h (original)
+++ trunk/include/asterisk/pbx.h Tue Dec 12 16:32:20 2006
@@ -156,7 +156,7 @@
* \param data the data passed into the app
*
* This application executes an application on a given channel. It
- * saves the stack and executes the given appliation passing in
+ * saves the stack and executes the given application passing in
* the given data.
*
* \return 0 on success, and -1 on failure
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