<description>
<para>Connects to the given TCP server, then transmits channel audio as 16-bit, 8KHz mono PCM over that socket (other codecs available via the channel driver interface). In turn, PCM audio is received from the socket and sent to the channel. Only audio frames and DTMF frames will be transmitted.</para>
<para>Protocol is specified at https://docs.asterisk.org/Configuration/Channel-Drivers/AudioSocket/</para>
- <para>This application does not automatically answer and should generally be preceeded by an application such as Answer() or Progress().</para>
+ <para>This application does not automatically answer and should generally be preceded by an application such as Answer() or Progress().</para>
</description>
</application>
***/
}
if (ast_audiosocket_init(svc, id)) {
- ast_log(LOG_ERROR, "Failed to intialize AudioSocket\n");
+ ast_log(LOG_ERROR, "Failed to initialize AudioSocket\n");
return -1;
}
const char *channel_name;
/*! Reset the CDR */
unsigned int reset:1;
- /*! If reseting the CDR, keep the variables */
+ /*! If resetting the CDR, keep the variables */
unsigned int keep_variables:1;
};
* \param user Optional Caller
* \param bridge_channel The bridged channel involved
*
- * \note if caller is NULL, the announcment will be sent to all participants in the conference.
+ * \note if caller is NULL, the announcement will be sent to all participants in the conference.
* \retval 0 on success.
* \retval -1 if the user hung up.
*/
nonexistent location, e.g. nonexistent DNS hostname.
</value>
<value name="CONGESTION">
- Channel or switching congestion occured when routing the call.
+ Channel or switching congestion occurred when routing the call.
This can occur if there is a slow or no response from the remote end.
</value>
<value name="NOANSWER">
channel back to itself. This will not echo CONTROL, MODEM, or NULL
frames. Note: If '#' detected application exits.</para>
<para>This application does not automatically answer and should be
- preceeded by an application such as Answer() or Progress().</para>
+ preceded by an application such as Answer() or Progress().</para>
</description>
</application>
***/
}
} else if (input[0] == EIVR_CMD_GET) {
char response[2048];
- ast_verb(4, "Retriving Variables from channel: %s\n", &input[2]);
+ ast_verb(4, "Retrieving Variables from channel: %s\n", &input[2]);
ast_eivr_getvariable(chan, &input[2], response, sizeof(response));
send_eivr_event(eivr_events, 'G', response, chan);
} else if (input[0] == EIVR_CMD_SVAR) {
ast_chan_log(LOG_NOTICE, chan, "Log message from EIVR: %s\n", &input[2]);
} else if (input[0] == EIVR_CMD_XIT) {
ast_chan_log(LOG_NOTICE, chan, "Exiting: %s\n", &input[2]);
- ast_chan_log(LOG_WARNING, chan, "e'X'it command is depricated, use 'E'xit instead\n");
+ ast_chan_log(LOG_WARNING, chan, "e'X'it command is deprecated, use 'E'xit instead\n");
res = 0;
break;
} else if (input[0] == EIVR_CMD_EXIT) {
* \param[in] jack_data This is the jack_data struct that contains the input
* ringbuffer that audio will be read from.
* \param[out] out_frame If this argument is non-NULL, then assuming there is
- * enough data avilable in the ringbuffer, the audio in this frame
+ * enough data available in the ringbuffer, the audio in this frame
* will get replaced with audio from the input buffer. If there is
* not enough data available to read at this time, then the frame
* data gets zeroed out.
<para>Total duration of conference in HH:MM:SS format.</para>
</parameter>
<parameter name="Creation">
- <para>How the conference was created: "Dyanmic" or "Static".</para>
+ <para>How the conference was created: "Dynamic" or "Static".</para>
</parameter>
<parameter name="Locked">
<para>Whether or not the conference is locked.</para>
<description>
<para>This application is part of the Mini-Voicemail system, configured in minivm.conf.</para>
<para>MinivmGreet() plays default prompts or user specific prompts for an account.</para>
- <para>Busy and unavailable messages can be choosen, but will be overridden if a temporary
+ <para>Busy and unavailable messages can be chosen, but will be overridden if a temporary
message exists for the account.</para>
<variablelist>
<variable name="MVM_GREET_STATUS">
}
/*!\internal
- * \brief Wraps a character sequence in double quotes, escaping occurences of quotes within the string.
+ * \brief Wraps a character sequence in double quotes, escaping occurrences of quotes within the string.
* \param from The string to work with.
* \param buf The destination buffer to write the modified quoted string.
* \param maxlen Always zero. \see ast_str
}
/*!\internal
- * \brief Send MWI using interal Asterisk event subsystem */
+ * \brief Send MWI using internal Asterisk event subsystem */
static int minivm_mwi_exec(struct ast_channel *chan, const char *data)
{
int argc;
return(load_config(1));
}
-/*! \brief Reload cofiguration */
+/*! \brief Reload configuration */
static char *handle_minivm_reload(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
</syntax>
<description>
<para>Records the audio on the current channel to the specified file.</para>
- <para>This application does not automatically answer and should be preceeded by
+ <para>This application does not automatically answer and should be preceded by
an application such as Answer or Progress().</para>
<note><para>MixMonitor runs as an audiohook.</para></note>
<note><para>If a filename passed to MixMonitor ends with
</syntax>
<description>
<para>Plays the Morse code equivalent of the passed string.</para>
- <para>This application does not automatically answer and should be preceeded by
+ <para>This application does not automatically answer and should be preceded by
an application such as Answer() or Progress().</para>
<para>This application uses the following variables:</para>
<variablelist>
<example title="Play an MP3 playlist">
exten => 1234,1,MP3Player(/var/lib/asterisk/playlist.m3u)
</example>
- <para>This application does not automatically answer and should be preceeded by an
+ <para>This application does not automatically answer and should be preceded by an
application such as Answer() or Progress().</para>
</description>
</application>
up by another user.</para>
<para>This application will return to the dialplan if the queue does not exist, or
any of the join options cause the caller to not enter the queue.</para>
- <para>This application does not automatically answer and should be preceeded
+ <para>This application does not automatically answer and should be preceded
by an application such as Answer(), Progress(), or Ringing().</para>
<para>This application sets the following channel variables upon completion:</para>
<variablelist>
<parameter name="queuename" required="true">
<enumlist>
<enum name="QUEUEMAX">
- <para>Maxmimum number of calls allowed.</para>
+ <para>Maximum number of calls allowed.</para>
</enum>
<enum name="QUEUESTRATEGY">
<para>The strategy of the queue.</para>
int periodicannouncestartdelay; /*!< How long into the queue should the periodic accouncement start */
int periodicannouncefrequency; /*!< How often to play periodic announcement */
int numperiodicannounce; /*!< The number of periodic announcements configured */
- int randomperiodicannounce; /*!< Are periodic announcments randomly chosen */
+ int randomperiodicannounce; /*!< Are periodic announcements randomly chosen */
int roundingseconds; /*!< How many seconds do we round to? */
int holdtime; /*!< Current avg holdtime, based on an exponential average */
int talktime; /*!< Current avg talktime, based on the same exponential average */
implement complicated spelling rules. We simply record the word
"men" in the sound file named "mans".</para>
<para>This application does not automatically answer and should be
- preceeded by an application such as Answer() or Progress.</para>
+ preceded by an application such as Answer() or Progress.</para>
</description>
<see-also>
<ref type="application">SayCountedAdj</ref>
SayCountedAdj(5,new,f) will play sound file "newx" (containing the
word "novikh").</para>
<para>This application does not automatically answer and should be
- preceeded by an application such as Answer(), Progress(), or
+ preceded by an application such as Answer(), Progress(), or
Proceeding().</para>
</description>
<see-also>
/*! \brief Find a skel level in the specified container
* \note This function *does not* look for a skel_level in the active container. It is used
* internally by the Config Options code to check if an level has already been added to the
- * container that will be swapped for the live container on a successul reload.
+ * container that will be swapped for the live container on a successful reload.
*
* \param tmp_container A non-active container to search for a level
* \param category The category associated with the level to check for
* bitfields in the config struct will have to use a custom handler
* \param opt The opaque config option
* \param var The ast_variable containing the option name and value
- * \param obj The object registerd for this option type
+ * \param obj The object registered for this option type
* \retval 0 Success
* \retval non-zero Failure
*/
);
if (!cfg) {
- ast_log(LOG_ERROR, "Couldn't access configuratino data!\n");
+ ast_log(LOG_ERROR, "Couldn't access configuration data!\n");
return -1;
}
S(SLA_TRUNK_STATE_ONHOLD)
S(SLA_TRUNK_STATE_ONHOLD_BYME)
}
- return "Uknown State";
+ return "Unknown State";
#undef S
}
<para>"Messages are processed as per text file message queues. smsq (a separate software) is a command to
generate message queues and send messages.</para>
<note><para>The protocol has tight delay bounds. Please use short frames and disable/keep short the
- jitter buffer on the ATA to make sure that respones (ACK etc.) are received in time.</para></note>
+ jitter buffer on the ATA to make sure that responses (ACK etc.) are received in time.</para></note>
</description>
</application>
***/
}
/*!
- * \brief Performs a change of the voicemail passowrd in the realtime engine.
+ * \brief Performs a change of the voicemail password in the realtime engine.
* \param vmu The voicemail user to change the password for.
* \param password The new value to be set to the password for this user.
*
* \param smsg the index of the message to be copied.
* \param ddir the destination folder to copy the message into.
* \param dmsg the index to be used for the copied message.
- * \param dmailboxuser The user who owns the mailbox tha contains the destination folder.
+ * \param dmailboxuser The user who owns the mailbox that contains the destination folder.
* \param dmailboxcontext The context for the destination user.
*
* This method is used for the COPY macro when mailboxes are stored in an ODBC back end.
* \param dmsg The destination message for the message to be renamed.
*
* This method is used by the RENAME macro when mailboxes are stored in an ODBC back end.
- * The is usually used to resequence the messages in the mailbox, such as to delete messag index 0, it would be called successively to slide all the other messages down one index.
+ * The is usually used to resequence the messages in the mailbox, such as to delete message index 0, it would be called successively to slide all the other messages down one index.
* But in theory, because the SQL query performs an update on (dir, msgnum, mailboxuser, mailboxcontext) in the database, it should be possible to have the message relocated to another mailbox or context as well.
*/
static void odbc_rename_message(char *sdir, int smsg, char *mailboxuser, char *mailboxcontext, char *ddir, int dmsg)
}
/*!
- * \brief Wraps a character sequence in double quotes, escaping occurences of quotes within the string.
+ * \brief Wraps a character sequence in double quotes, escaping occurrences of quotes within the string.
* \param from The string to work with.
* \param buf The buffer into which to write the modified quoted string.
* \param maxlen Always zero, but see \see ast_str
/*!
* \brief Variables used for saving a voicemail.
*
- * This includes the record gain, mode flags, and the exit context of the chanel that was used for leaving the voicemail.
+ * This includes the record gain, mode flags, and the exit context of the channel that was used for leaving the voicemail.
*/
struct leave_vm_options {
unsigned int flags;
smdi_iface = ast_smdi_interface_find("/dev/ttyS0");
}
if (!smdi_iface) {
- ast_log(AST_LOG_ERROR, "No valid SMDI interface specfied, disabling SMDI voicemail notification\n");
+ ast_log(AST_LOG_ERROR, "No valid SMDI interface specified, disabling SMDI voicemail notification\n");
}
}
ast_copy_string(vminfo_args, test_items[test_counter].vminfo_test_args, sizeof(vminfo_args));
test_ret = acf_vm_info(chan, vminfo_cmd, vminfo_args, vminfo_buf, sizeof(vminfo_buf));
if (strcmp(vminfo_buf, test_items[test_counter].vminfo_expected)) {
- ast_test_status_update(test, "VM_INFO respose was: '%s', but expected: '%s'\n", vminfo_buf, test_items[test_counter].vminfo_expected);
+ ast_test_status_update(test, "VM_INFO response was: '%s', but expected: '%s'\n", vminfo_buf, test_items[test_counter].vminfo_expected);
res = AST_TEST_FAIL;
}
if (!(test_ret == test_items[test_counter].vminfo_ret)) {
* \author David C. Troy <dave@popvox.com>
*
* \brief Wait For Noise
- * The same as Wait For Silence but listenes noise on the chennel that is above \n
+ * The same as Wait For Silence but listens for noise on the channel that is above \n
* the pre-configured silence threshold from dsp.conf
*
* \author Philipp Skadorov <skadorov@yahoo.com>
conference to a specific number. By default conferences
have no participant limit. After the limit is reached, the
conference will be locked until someone leaves. Note however
- that an Admin user will always be alowed to join the conference
+ that an Admin user will always be allowed to join the conference
regardless if this limit is reached or not.
</para></description>
</configOption>
<enum name="playback(filename&filename2&...)"><para>
<literal>playback</literal> will play back an audio file to a channel
and then immediately return to the conference.
- This file can not be interupted by DTMF.
+ This file can not be interrupted by DTMF.
Multiple files can be chained together using the
<literal>&</literal> character.</para></enum>
<enum name="playback_and_continue(filename&filename2&...)"><para>
BRIDGE_OPT_REMB_BEHAVIOR_AVERAGE_ALL = (1 << 12), /*!< The average of all REMB reports in the entire bridge is sent to each sender */
BRIDGE_OPT_REMB_BEHAVIOR_LOWEST_ALL = (1 << 13), /*!< The lowest estimated maximum bitrate from all receivers is sent to each sender */
BRIDGE_OPT_REMB_BEHAVIOR_HIGHEST_ALL = (1 << 14), /*!< The highest estimated maximum bitrate from all receivers is sent to each sender */
- BRIDGE_OPT_REMB_BEHAVIOR_FORCE = (1 << 15), /*!< Force the REMB estimated bitrate to that specifiec in remb_estimated_bitrate */
+ BRIDGE_OPT_REMB_BEHAVIOR_FORCE = (1 << 15), /*!< Force the REMB estimated bitrate to that specified in remb_estimated_bitrate */
};
enum conf_menu_action_id {
projects / thirdparty / asterisk.git / commitdiff
