diff --git a/ChangeLog b/ChangeLog
index d336a2a..c5ec22a 100644
--- a/ChangeLog
+++ b/ChangeLog
@@ -14722,7 +14722,7 @@
 	  peer's allowtransfer setting The pvt_sip allowtransfer was not
 	  being set to that of the peer's setting. Therefore, the global
 	  allowtransfer setting was being used instead which would lead to
-	  calls not being transfered if the global setting was set to 'no'
+	  calls not being transferred if the global setting was set to 'no'
 	  despite the setting on the peer being 'yes' and vice versa, calls
 	  would be allowed to transfer even if the peer's setting was 'no'
 	  but the global setting was 'yes'. (Closes issue ASTERISK-19856)
@@ -18131,7 +18131,7 @@
 	* apps/app_queue.c: Adds a transfer callee on hangup option (like
 	  with Dial option F) to queues. This should (and does in my
 	  testing) act just like the Dial option of the same name. This
-	  allows a queue member to be transfered to the next priority (no
+	  allows a queue member to be transferred to the next priority (no
 	  args), or to a context/extension/priority similar to goto (with
 	  args context^extension^priority) when a caller hangs up on them.
 	  (closes issue ASTERISK-19283) Reported by: To Patches:
diff --git a/apps/app_transfer.c b/apps/app_transfer.c
index 9dc2ab2..255857d 100644
--- a/apps/app_transfer.c
+++ b/apps/app_transfer.c
@@ -54,7 +54,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision: 361042 $")
 		<description>
 			<para>Requests the remote caller be transferred
 			to a given destination. If TECH (SIP, IAX2, LOCAL etc) is used, only
-			an incoming call with the same channel technology will be transfered.
+			an incoming call with the same channel technology will be transferred.
 			Note that for SIP, if you transfer before call is setup, a 302 redirect
 			SIP message will be returned to the caller.</para>
 			<para>The result of the application will be reported in the <variable>TRANSFERSTATUS</variable>
diff --git a/channels/chan_iax2.c b/channels/chan_iax2.c
index cef349c..32d7a72 100644
--- a/channels/chan_iax2.c
+++ b/channels/chan_iax2.c
@@ -5602,7 +5602,7 @@ static enum ast_bridge_result iax2_bridge(struct ast_channel *c0, struct ast_cha
 			unlock_both(callno0, callno1);
 			return AST_BRIDGE_FAILED_NOWARN;
 		}
-		/* check if transfered and if we really want native bridging */
+		/* check if transferred and if we really want native bridging */
 		if (!transferstarted && !ast_test_flag64(iaxs[callno0], IAX_NOTRANSFER) && !ast_test_flag64(iaxs[callno1], IAX_NOTRANSFER)) {
 			/* Try the transfer */
 			if (iax2_start_transfer(callno0, callno1, (flags & (AST_BRIDGE_DTMF_CHANNEL_0 | AST_BRIDGE_DTMF_CHANNEL_1)) ||
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 4fc2322..1da0f21 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -6290,7 +6290,7 @@ static int sip_call(struct ast_channel *ast, const char *dest, int timeout)
 		} else if (!strcasecmp(ast_var_name(current), "SIPFROMDOMAIN")) {
 			ast_string_field_set(p, fromdomain, ast_var_value(current));
 		} else if (!strcasecmp(ast_var_name(current), "SIPTRANSFER")) {
-			/* This is a transfered call */
+			/* This is a transferred call */
 			p->options->transfer = 1;
 		} else if (!strcasecmp(ast_var_name(current), "SIPTRANSFER_REFERER")) {
 			/* This is the referrer */
@@ -6347,7 +6347,7 @@ static int sip_call(struct ast_channel *ast, const char *dest, int timeout)
 
 		if (referer) {
 			if (sipdebug)
-				ast_debug(3, "Call for %s transfered by %s\n", p->username, referer);
+				ast_debug(3, "Call for %s transferred by %s\n", p->username, referer);
 			snprintf(buf, sizeof(buf)-1, "-> %s (via %s)", p->cid_name, referer);
 		} else
 			snprintf(buf, sizeof(buf)-1, "-> %s", p->cid_name);
@@ -24514,7 +24514,7 @@ static int handle_request_notify(struct sip_pvt *p, struct sip_request *req, str
 
 		/* EventID for each transfer... EventID is basically the REFER cseq
 
-		 We are getting notifications on a call that we transfered
+		 We are getting notifications on a call that we transferred
 		 We should hangup when we are getting a 200 OK in a sipfrag
 		 Check if we have an owner of this event */
 
diff --git a/main/features.c b/main/features.c
index dbc84c4..126ebcc 100644
--- a/main/features.c
+++ b/main/features.c
@@ -2497,7 +2497,7 @@ static const char *real_ctx(struct ast_channel *transferer, struct ast_channel *
 
 /*!
  * \brief Blind transfer user to another extension
- * \param chan channel to be transfered
+ * \param chan channel to be transferred
  * \param peer channel initiated blind transfer
  * \param config
  * \param code
@@ -2661,7 +2661,7 @@ static void atxfer_fail_cleanup(struct ast_channel *transferee, struct ast_chann
 
 /*!
  * \brief Attended transfer
- * \param chan transfered user
+ * \param chan transferred user
  * \param peer person transfering call
  * \param config
  * \param code
@@ -3845,7 +3845,7 @@ static void set_config_flags(struct ast_channel *chan, struct ast_bridge_config
  * \details
  * Request channel, set channel variables, initiate call,
  * check if they want to disconnect, go into loop, check if timeout has elapsed,
- * check if person to be transfered hung up, check for answer break loop,
+ * check if person to be transferred hung up, check for answer break loop,
  * set cdr return channel.
  *
  * \retval Channel Connected channel for transfer.
@@ -4463,7 +4463,7 @@ int ast_bridge_call(struct ast_channel *chan, struct ast_channel *peer, struct a
 				ast_set_flag(peer_cdr, AST_CDR_FLAG_BRIDGED);
 			}
 		}
-		/* the DIALED flag may be set if a dialed channel is transfered
+		/* the DIALED flag may be set if a dialed channel is transferred
 		 * and then bridged to another channel.  In order for the
 		 * bridge CDR to be written, the DIALED flag must not be
 		 * present. */
@@ -8157,7 +8157,7 @@ int ast_bridge_timelimit(struct ast_channel *chan, struct ast_bridge_config *con
  *
  * Split data, check we aren't bridging with ourself, check valid channel,
  * answer call if not already, check compatible channels, setup bridge config
- * now bridge call, if transfered party hangs up return to PBX extension.
+ * now bridge call, if transferred party hangs up return to PBX extension.
  */
 static int bridge_exec(struct ast_channel *chan, const char *data)
 {
diff --git a/main/manager.c b/main/manager.c
index fc3d4e9..da4437f 100644
--- a/main/manager.c
+++ b/main/manager.c
@@ -4498,7 +4498,7 @@ static int action_originate(struct mansession *s, const struct message *m)
 			ast_string_field_set(fast, account, account);
 			fast->vars = vars;
 			fast->cap = cap;
-			cap = NULL; /* transfered originate helper the capabilities structure.  It is now responsible for freeing it. */
+			cap = NULL; /* transferred originate helper the capabilities structure.  It is now responsible for freeing it. */
 			fast->timeout = to;
 			fast->early_media = bridge_early;
 			fast->priority = pi;