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

[Josh Soref]

Josh Soref has uploaded this change for review. ( https://gerrit.asterisk.org/c/asterisk/+/17118 )


Change subject: spelling: probably
......................................................................

spelling: probably

Change-Id: I24c9fe9692db72f185ccb8306df0b4eab143cf37
---
M UPGRADE.txt
M channels/chan_sip.c
M include/asterisk/callerid.h
3 files changed, 4 insertions(+), 4 deletions(-)



  git pull ssh://gerrit.asterisk.org:29418/asterisk refs/changes/18/17118/1

diff --git a/UPGRADE.txt b/UPGRADE.txt
index 3ad163a..5cbbfee 100644
--- a/UPGRADE.txt
+++ b/UPGRADE.txt
@@ -2600,7 +2600,7 @@
 
 * The "canreinvite" option has changed. canreinvite=yes used to disable
   re-invites if you had NAT=yes. In 1.4, you need to set canreinvite=nonat
-  to disable re-invites when NAT=yes. This is propably what you want.
+  to disable re-invites when NAT=yes. This is probably what you want.
   The settings are now: "yes", "no", "nonat", "update". Please consult
   sip.conf.sample for detailed information.
 
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 654fdfd..eb9d4f1 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -114,7 +114,7 @@
  * \todo Add TCP/TLS information to function SIPPEER and CHANNEL function
  * \todo If tcpenable=yes, we must open a TCP socket on the same address as the IP for UDP.
  * 	The tcpbindaddr config option should only be used to open ADDITIONAL ports
- * 	So we should propably go back to
+ * 	So we should probably go back to
  *		bindaddr= the default address to bind to. If tcpenable=yes, then bind this to both udp and TCP
  *				if tlsenable=yes, open TLS port (provided we also have cert)
  *		tcpbindaddr = extra address for additional TCP connections
@@ -16452,7 +16452,7 @@
 	p->refer->status = REFER_SENT;   /* Set refer status */
 
 	return transmit_invite(p, SIP_REFER, FALSE, 0, NULL);
-	/* We should propably wait for a NOTIFY here until we ack the transfer */
+	/* We should probably wait for a NOTIFY here until we ack the transfer */
 	/* Maybe fork a new thread and wait for a STATUS of REFER_200OK on the refer status before returning to app_transfer */
 
 	/*! \todo In theory, we should hang around and wait for a reply, before
diff --git a/include/asterisk/callerid.h b/include/asterisk/callerid.h
index 777553d..7788aca 100644
--- a/include/asterisk/callerid.h
+++ b/include/asterisk/callerid.h
@@ -27,7 +27,7 @@
 /*!
  * \page CID Caller ID names and numbers
  *
- * Caller ID names are currently 8 bit characters, propably
+ * Caller ID names are currently 8 bit characters, probably
  * ISO8859-1, depending on what your channel drivers handle.
  *
  * IAX2 and SIP caller ID names are UTF8

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Gerrit-Project: asterisk
Gerrit-Branch: master
Gerrit-Change-Id: I24c9fe9692db72f185ccb8306df0b4eab143cf37
Gerrit-Change-Number: 17118
Gerrit-PatchSet: 1
Gerrit-Owner: Josh Soref <jsoref at gmail.com>
Gerrit-CC: Friendly Automation
Gerrit-MessageType: newchange
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