- fix typo in rtp.c, devicestate.h

- add information about subscriptions and realtime dial plans in sip.conf.sample

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@24342 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample
index 26966cd..84200a5 100644 (file)
--- a/configs/sip.conf.sample
+++ b/configs/sip.conf.sample
@@ -29,7 +29,6 @@ context=default                       ; Default context for incoming calls
                                ; this can also be set to 'osp'
                                ; if asterisk was compiled with OSP support.)
 allowoverlap=no                        ; Disable overlap dialing support. (Default is yes)
-;allowsubscribe=no             ; Disable support for subscriptions. (Default is yes)
 ;realm=mydomain.tld            ; Realm for digest authentication
                                ; defaults to "asterisk"
                                ; Realms MUST be globally unique according to RFC 3261
@@ -114,10 +113,6 @@ srvlookup=yes                      ; Enable DNS SRV lookups on outbound calls
 ;compactheaders = yes          ; send compact sip headers.
 ;sipdebug = yes                        ; Turn on SIP debugging by default, from
                                ; the moment the channel loads this configuration
-;subscribecontext = default    ; Set a specific context for SUBSCRIBE requests
-                               ; Useful to limit subscriptions to local extensions
-                               ; Settable per peer/user also
-;notifyringing = yes           ; Notify subscriptions on RINGING state
 ;
 ;videosupport=yes              ; Turn on support for SIP video
 ;maxcallbitrate=384            ; Maximum bitrate for video calls (default 384 kb/s)
@@ -126,6 +121,18 @@ srvlookup=yes                      ; Enable DNS SRV lookups on outbound calls
 ;callevents=no                 ; generate manager events when sip ua 
                                ; performs events (e.g. hold)
 
+;--------------------------- STATUS NOTIFICATIONS (SUBSCRIPTIONS) ---------
+; You can subscribe to the status of extensions with a "hint" priority
+; (See extensions.conf.sample for examples)
+; chan_sip support two major formats for notifications: dialog-info and SIMPLE 
+; Note: Subscriptions does not work if you have a realtime dialplan and use the
+; realtime switch.
+;
+;allowsubscribe=no             ; Disable support for subscriptions. (Default is yes)
+;subscribecontext = default    ; Set a specific context for SUBSCRIBE requests
+                               ; Useful to limit subscriptions to local extensions
+                               ; Settable per peer/user also
+;notifyringing = yes           ; Notify subscriptions on RINGING state
 ;
 ; If regcontext is specified, Asterisk will dynamically create and destroy a
 ; NoOp priority 1 extension for a given peer who registers or unregisters with

diff --git a/include/asterisk/devicestate.h b/include/asterisk/devicestate.h
index e41b9ee..1892ab5 100644 (file)
--- a/include/asterisk/devicestate.h
+++ b/include/asterisk/devicestate.h
@@ -79,7 +79,7 @@ int ast_device_state_changed(const char *fmt, ...)
 
 
 /*! \brief Tells Asterisk the State for Device is changed 
- * \param device devicename like a dialstrin
+ * \param device devicename like a dialstring
  * Asterisk polls the new extensionstates and calls the registered
  * callbacks for the changed extensions
  * Returns 0 on success, -1 on failure

diff --git a/rtp.c b/rtp.c
index e55c379..88aa1ee 100644 (file)
--- a/rtp.c
+++ b/rtp.c
@@ -23,7 +23,7 @@
  *
  * \author Mark Spencer <markster@digium.com>
  * 
- * \note RTP is deffined in RFC 3550.
+ * \note RTP is defined in RFC 3550.
  */
 
 #include <stdio.h>