[asterisk-commits] rizzo: branch rizzo/astobj2 r51224 -
/team/rizzo/astobj2/channels/chan_sip.c
asterisk-commits at lists.digium.com
asterisk-commits at lists.digium.com
Thu Jan 18 09:27:43 MST 2007
Author: rizzo
Date: Thu Jan 18 10:27:42 2007
New Revision: 51224
URL: http://svn.digium.com/view/asterisk?view=rev&rev=51224
Log:
merge from trunk 48417 - typos in comments
Modified:
team/rizzo/astobj2/channels/chan_sip.c
Modified: team/rizzo/astobj2/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/team/rizzo/astobj2/channels/chan_sip.c?view=diff&rev=51224&r1=51223&r2=51224
==============================================================================
--- team/rizzo/astobj2/channels/chan_sip.c (original)
+++ team/rizzo/astobj2/channels/chan_sip.c Thu Jan 18 10:27:42 2007
@@ -851,7 +851,7 @@
#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)
@@ -881,7 +881,7 @@
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 */
@@ -1299,7 +1299,7 @@
int refresh; /*!< How often to refresh */
struct sip_pvt *register_pvt; /*!< 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 */
@@ -3145,7 +3145,7 @@
struct sip_pvt *p = ast->tech_pvt; /* chan is locked, no need to grab reference */
struct varshead *headp;
struct ast_var_t *current;
- const char *referer = NULL; /* SIP refererer */
+ const char *referer = NULL; /* SIP referrer */
if ((ast->_state != AST_STATE_DOWN) && (ast->_state != AST_STATE_RESERVED)) {
ast_log(LOG_WARNING, "sip_call called on %s, neither down nor reserved\n", ast->name);
@@ -3167,7 +3167,7 @@
/* 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. */
@@ -3385,9 +3385,9 @@
* 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
*
*/
@@ -3539,7 +3539,7 @@
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;
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