[Asterisk-code-review] spelling: available (asterisk[master])

[Josh Soref]

Josh Soref has uploaded this change for review. ( https://gerrit.asterisk.org/c/asterisk/+/16759 )


Change subject: spelling: available
......................................................................

spelling: available

Change-Id: I320978b3ca2e72e3d0fb469b0a68c7651e96e8de
---
M CHANGES
M apps/app_jack.c
M apps/app_queue.c
M cdr/cdr_beanstalkd.c
M cel/cel_beanstalkd.c
M channels/chan_dahdi.c
M main/pbx.c
7 files changed, 7 insertions(+), 7 deletions(-)



  git pull ssh://gerrit.asterisk.org:29418/asterisk refs/changes/59/16759/1

diff --git a/CHANGES b/CHANGES
index e4400c2..05ec24e 100644
--- a/CHANGES
+++ b/CHANGES
@@ -926,7 +926,7 @@
 
 res_rtp_asterisk
 ------------------
- * The existing strictrtp option in rtp.conf has a new choice availabe, called
+ * The existing strictrtp option in rtp.conf has a new choice available, called
    'seqno', which behaves the same way as setting strictrtp to 'yes', but will
    ignore the time interval during learning so that bursts of packets can still
    trigger learning our source.
diff --git a/apps/app_jack.c b/apps/app_jack.c
index 1609972..1abe490 100644
--- a/apps/app_jack.c
+++ b/apps/app_jack.c
@@ -607,7 +607,7 @@
  * \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.
diff --git a/apps/app_queue.c b/apps/app_queue.c
index b0f463b..b15d3cf 100644
--- a/apps/app_queue.c
+++ b/apps/app_queue.c
@@ -5576,7 +5576,7 @@
 		res = 0;
 	}
 
-	/* Update realtime members if this is the first call and number of avalable members is 0 */
+	/* Update realtime members if this is the first call and number of available members is 0 */
 	if (avl == 0 && qe->pos == 1) {
 		update_realtime_members(qe->parent);
 	}
diff --git a/cdr/cdr_beanstalkd.c b/cdr/cdr_beanstalkd.c
index 524274f..107cdfc 100644
--- a/cdr/cdr_beanstalkd.c
+++ b/cdr/cdr_beanstalkd.c
@@ -20,7 +20,7 @@
  * \file
  * \brief Asterisk Beanstalkd CDR records.
  *
- * This module requires the beanstalk-client library, avaialble from
+ * This module requires the beanstalk-client library, available from
  * https://github.com/deepfryed/beanstalk-client
  *
  * See also
diff --git a/cel/cel_beanstalkd.c b/cel/cel_beanstalkd.c
index fe4f430..1bfe2e8 100644
--- a/cel/cel_beanstalkd.c
+++ b/cel/cel_beanstalkd.c
@@ -22,7 +22,7 @@
  *
  * \brief Asterisk Channel Event Beanstalkd backend
  *
- * This module requires the beanstalk-client library, avaialble from
+ * This module requires the beanstalk-client library, available from
  * https://github.com/deepfryed/beanstalk-client
  *
  * See also
diff --git a/channels/chan_dahdi.c b/channels/chan_dahdi.c
index 74c76f7..e750648 100644
--- a/channels/chan_dahdi.c
+++ b/channels/chan_dahdi.c
@@ -6234,7 +6234,7 @@
 				p->subs[SUB_REAL].inthreeway = 0;
 			}
 		} else if (idx == SUB_CALLWAIT) {
-			/* Ditch the holding callwait call, and immediately make it availabe */
+			/* Ditch the holding callwait call, and immediately make it available */
 			if (p->subs[SUB_CALLWAIT].inthreeway) {
 				/* This is actually part of a three way, placed on hold.  Place the third part
 				   on music on hold now */
diff --git a/main/pbx.c b/main/pbx.c
index d0ee127..dbab50e 100644
--- a/main/pbx.c
+++ b/main/pbx.c
@@ -4330,7 +4330,7 @@
 	callid = ast_read_threadstorage_callid();
 	/* If the thread isn't already associated with a callid, we should create that association. */
 	if (!callid) {
-		/* Associate new PBX thread with the channel call id if it is availble.
+		/* Associate new PBX thread with the channel call id if it is available.
 		 * If not, create a new one instead.
 		 */
 		callid = ast_channel_callid(c);

-- 
To view, visit https://gerrit.asterisk.org/c/asterisk/+/16759
To unsubscribe, or for help writing mail filters, visit https://gerrit.asterisk.org/settings

Gerrit-Project: asterisk
Gerrit-Branch: master
Gerrit-Change-Id: I320978b3ca2e72e3d0fb469b0a68c7651e96e8de
Gerrit-Change-Number: 16759
Gerrit-PatchSet: 1
Gerrit-Owner: Josh Soref <jsoref at gmail.com>
Gerrit-CC: Friendly Automation
Gerrit-MessageType: newchange
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-code-review/attachments/20211107/9aaf6a01/attachment-0001.html>