[asterisk-commits] trunk r24342 - in /trunk: ./ configs/ include/asterisk/

[asterisk-commits at lists.digium.com]

Author: oej
Date: Tue May  2 15:31:39 2006
New Revision: 24342

URL: http://svn.digium.com/view/asterisk?rev=24342&view=rev
Log:
- fix typo in rtp.c, devicestate.h
- add information about subscriptions and realtime dial plans in sip.conf.sample

Modified:
    trunk/configs/sip.conf.sample
    trunk/include/asterisk/devicestate.h
    trunk/rtp.c

Modified: trunk/configs/sip.conf.sample
URL: http://svn.digium.com/view/asterisk/trunk/configs/sip.conf.sample?rev=24342&r1=24341&r2=24342&view=diff
==============================================================================
--- trunk/configs/sip.conf.sample (original)
+++ trunk/configs/sip.conf.sample Tue May  2 15:31:39 2006
@@ -29,7 +29,6 @@
 				; 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 @@
 ;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 @@
 ;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

Modified: trunk/include/asterisk/devicestate.h
URL: http://svn.digium.com/view/asterisk/trunk/include/asterisk/devicestate.h?rev=24342&r1=24341&r2=24342&view=diff
==============================================================================
--- trunk/include/asterisk/devicestate.h (original)
+++ trunk/include/asterisk/devicestate.h Tue May  2 15:31:39 2006
@@ -79,7 +79,7 @@
 
 
 /*! \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

Modified: trunk/rtp.c
URL: http://svn.digium.com/view/asterisk/trunk/rtp.c?rev=24342&r1=24341&r2=24342&view=diff
==============================================================================
--- trunk/rtp.c (original)
+++ trunk/rtp.c Tue May  2 15:31:39 2006
@@ -23,7 +23,7 @@
  *
  * \author Mark Spencer <markster at digium.com>
  * 
- * \note RTP is deffined in RFC 3550.
+ * \note RTP is defined in RFC 3550.
  */
 
 #include <stdio.h>