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Several components: fixing Typos in comments and code, "avaliable" instead of "available" 

(issue ASTERISK-23021)
(closes issue ASTERISK-23021)
Reported by: Jeremy Lainé
Tested by: Rusty Newton
Patches:
   available.patch uploaded by Jeremy Lainé (license 6561)
........

Merged revisions 404046 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404047 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Rusty Newton 2013-12-17 23:38:02 +00:00
parent b0bb03e916
commit f7c60b8fb6
6 changed files with 6 additions and 6 deletions
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2
apps/app_chanspy.c
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@ -372,7 +372,7 @@ enum {
OPTION_NAME = (1 << 12), /* Say the name of the person on whom we will spy */
OPTION_DTMF_SWITCH_MODES = (1 << 13), /* Allow numeric DTMF to switch between chanspy modes */
OPTION_DTMF_EXIT = (1 << 14), /* Set DTMF to exit, added for DAHDIScan integration */
OPTION_DTMF_CYCLE = (1 << 15), /* Custom DTMF for cycling next avaliable channel, (default is '*') */
OPTION_DTMF_CYCLE = (1 << 15), /* Custom DTMF for cycling next available channel, (default is '*') */
OPTION_DAHDI_SCAN = (1 << 16), /* Scan groups in DAHDIScan mode */
OPTION_STOP = (1 << 17),
OPTION_EXITONHANGUP = (1 << 18), /* Hang up when the spied-on channel hangs up. */

2
apps/app_mixmonitor.c
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@ -1101,7 +1101,7 @@ static int stop_mixmonitor_full(struct ast_channel *chan, const char *data)
ast_mutex_lock(&mixmonitor_ds->lock);
/* closing the filestream here guarantees the file is avaliable to the dialplan
/* closing the filestream here guarantees the file is available to the dialplan
* after calling StopMixMonitor */
mixmonitor_ds_close_fs(mixmonitor_ds);

2
channels/chan_iax2.c
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@ -2871,7 +2871,7 @@ static int create_callno_pools(void)
* \brief Schedules delayed removal of iax2_pvt call number data
*
* \note After MIN_REUSE_TIME has passed for a destroyed iax2_pvt, the callno is
* avaliable again, and the address from the previous connection must be decremented
* available again, and the address from the previous connection must be decremented
* from the peercnts table. This function schedules these operations to take place.
*/
static void sched_delay_remove(struct ast_sockaddr *addr, callno_entry entry)

2
include/asterisk/test.h
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@ -45,7 +45,7 @@
Create a callback function for the test using the AST_TEST_DEFINE macro.
Each defined test has three arguments avaliable to it's test code.
Each defined test has three arguments available to it's test code.
\param struct ast_test_info *info
\param enum ast_test_command cmd
\param struct ast_test *test

2
main/channel.c
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@ -5626,7 +5626,7 @@ struct ast_channel *ast_call_forward(struct ast_channel *caller, struct ast_chan
ast_channel_accountcode_set(new_chan, oh->account);
ast_channel_unlock(new_chan);
}
} else if (caller) { /* no outgoing helper so use caller if avaliable */
} else if (caller) { /* no outgoing helper so use caller if available */
call_forward_inherit(new_chan, caller, orig);
}
ast_set_flag(ast_channel_flags(new_chan), AST_FLAG_ORIGINATED);

2
main/rtp_engine.c
View File

@ -1610,7 +1610,7 @@ static void add_static_payload(int map, const struct ast_format *format, int rtp
}
if (map < 0) {
ast_log(LOG_WARNING, "No Dynamic RTP mapping avaliable for format %s\n" ,ast_getformatname(format));
ast_log(LOG_WARNING, "No Dynamic RTP mapping available for format %s\n" ,ast_getformatname(format));
ast_rwlock_unlock(&static_RTP_PT_lock);
return;
}