[asterisk-commits] lathama: trunk r305843 - in /trunk: ./ apps/ configs/ funcs/

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Wed Feb 2 13:30:53 CST 2011


Author: lathama
Date: Wed Feb  2 13:30:49 2011
New Revision: 305843

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=305843
Log:
Replacing doc/* and asterisk.pdf with wiki links

Adding links to http(s)://wiki.asterisk.org


Modified:
    trunk/   (props changed)
    trunk/apps/app_externalivr.c
    trunk/apps/app_voicemail.c
    trunk/configs/dundi.conf.sample
    trunk/configs/h323.conf.sample
    trunk/configs/iax.conf.sample
    trunk/configs/iaxprov.conf.sample
    trunk/configs/mgcp.conf.sample
    trunk/configs/sip.conf.sample
    trunk/configs/skinny.conf.sample
    trunk/configs/sla.conf.sample
    trunk/configs/unistim.conf.sample
    trunk/funcs/func_callcompletion.c
    trunk/funcs/func_enum.c

Propchange: trunk/
------------------------------------------------------------------------------
Binary property 'branch-1.8-merged' - no diff available.

Modified: trunk/apps/app_externalivr.c
URL: http://svnview.digium.com/svn/asterisk/trunk/apps/app_externalivr.c?view=diff&rev=305843&r1=305842&r2=305843
==============================================================================
--- trunk/apps/app_externalivr.c (original)
+++ trunk/apps/app_externalivr.c Wed Feb  2 13:30:49 2011
@@ -85,7 +85,7 @@
 			all DTMF events received on the channel, and notification if the channel is
 			hung up. The received on the channel, and notification if the channel is hung
 			up. The application will not be forcibly terminated when the channel is hung up.
-			For more information see <filename>doc/asterisk.pdf</filename>.</para>
+			For more information see <filename>doc/AST.pdf</filename>.</para>
 		</description>
 	</application>
  ***/

Modified: trunk/apps/app_voicemail.c
URL: http://svnview.digium.com/svn/asterisk/trunk/apps/app_voicemail.c?view=diff&rev=305843&r1=305842&r2=305843
==============================================================================
--- trunk/apps/app_voicemail.c (original)
+++ trunk/apps/app_voicemail.c Wed Feb  2 13:30:49 2011
@@ -27,7 +27,7 @@
  *
  * \par See also
  * \arg \ref Config_vm
- * \note For information about voicemail IMAP storage, read doc/asterisk.pdf
+ * \note For information about voicemail IMAP storage, https://wiki.asterisk.org/wiki/display/AST/IMAP+Voicemail+Storage
  * \ingroup applications
  * \note This module requires res_adsi to load. This needs to be optional
  * during compilation.

Modified: trunk/configs/dundi.conf.sample
URL: http://svnview.digium.com/svn/asterisk/trunk/configs/dundi.conf.sample?view=diff&rev=305843&r1=305842&r2=305843
==============================================================================
--- trunk/configs/dundi.conf.sample (original)
+++ trunk/configs/dundi.conf.sample Wed Feb  2 13:30:49 2011
@@ -27,7 +27,7 @@
 ;bindaddr=0.0.0.0
 ;port=4520
 ;
-; See qos.tex or Quality of Service section of asterisk.pdf for a description of the tos parameter.
+; See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for a description of the tos parameter.
 ;tos=ef
 ;
 ; Our entity identifier (Should generally be the MAC address of the

Modified: trunk/configs/h323.conf.sample
URL: http://svnview.digium.com/svn/asterisk/trunk/configs/h323.conf.sample?view=diff&rev=305843&r1=305842&r2=305843
==============================================================================
--- trunk/configs/h323.conf.sample (original)
+++ trunk/configs/h323.conf.sample Wed Feb  2 13:30:49 2011
@@ -5,7 +5,7 @@
 port = 1720
 ;bindaddr = 1.2.3.4 	; this SHALL contain a single, valid IP address for this machine
 ;
-; See qos.tex or Quality of Service section of asterisk.pdf for a description of these parameters.
+; See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for a description of these parameters.
 ;tos_audio=ef		; Sets TOS for RTP audio packets.
 ;cos_audio=5		; Sets 802.1p priority for RTP audio packets.
 ;

Modified: trunk/configs/iax.conf.sample
URL: http://svnview.digium.com/svn/asterisk/trunk/configs/iax.conf.sample?view=diff&rev=305843&r1=305842&r2=305843
==============================================================================
--- trunk/configs/iax.conf.sample (original)
+++ trunk/configs/iax.conf.sample Wed Feb  2 13:30:49 2011
@@ -259,7 +259,7 @@
 ;
 ;authdebug=no
 ;
-; See qos.tex or Quality of Service section of asterisk.pdf for a description of these parameters.
+; See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for a description of these parameters.
 ;tos=ef
 ;cos=5
 ;

Modified: trunk/configs/iaxprov.conf.sample
URL: http://svnview.digium.com/svn/asterisk/trunk/configs/iaxprov.conf.sample?view=diff&rev=305843&r1=305842&r2=305843
==============================================================================
--- trunk/configs/iaxprov.conf.sample (original)
+++ trunk/configs/iaxprov.conf.sample Wed Feb  2 13:30:49 2011
@@ -53,7 +53,7 @@
 ;
 flags=register,heartbeat
 ;
-; See qos.tex or Quality of Service section of asterisk.pdf for a description of this parameter.
+; See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for a description of this parameter.
 ;tos=ef
 ;
 ; Example iaxy provisioning

Modified: trunk/configs/mgcp.conf.sample
URL: http://svnview.digium.com/svn/asterisk/trunk/configs/mgcp.conf.sample?view=diff&rev=305843&r1=305842&r2=305843
==============================================================================
--- trunk/configs/mgcp.conf.sample (original)
+++ trunk/configs/mgcp.conf.sample Wed Feb  2 13:30:49 2011
@@ -5,7 +5,7 @@
 ;port = 2427
 ;bindaddr = 0.0.0.0
 
-; See qos.tex or Quality of Service section of asterisk.pdf for a description of these parameters.
+; See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for a description of these parameters.
 ;tos=cs3		; Sets TOS for signaling packets.
 ;tos_audio=ef		; Sets TOS for RTP audio packets.
 ;cos=3			; Sets 802.1p priority for signaling packets.

Modified: trunk/configs/sip.conf.sample
URL: http://svnview.digium.com/svn/asterisk/trunk/configs/sip.conf.sample?view=diff&rev=305843&r1=305842&r2=305843
==============================================================================
--- trunk/configs/sip.conf.sample (original)
+++ trunk/configs/sip.conf.sample Wed Feb  2 13:30:49 2011
@@ -217,7 +217,7 @@
                                 ; and multiline formatted headers for strict
                                 ; SIP compatibility (defaults to "yes")
 
-; See qos.tex or Quality of Service section of asterisk.pdf for a description of these parameters.
+; See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for a description of these parameters.
 ;tos_sip=cs3                    ; Sets TOS for SIP packets.
 ;tos_audio=ef                   ; Sets TOS for RTP audio packets.
 ;tos_video=af41                 ; Sets TOS for RTP video packets.

Modified: trunk/configs/skinny.conf.sample
URL: http://svnview.digium.com/svn/asterisk/trunk/configs/skinny.conf.sample?view=diff&rev=305843&r1=305842&r2=305843
==============================================================================
--- trunk/configs/skinny.conf.sample (original)
+++ trunk/configs/skinny.conf.sample Wed Feb  2 13:30:49 2011
@@ -29,7 +29,7 @@
 			; for framing options
 ;disallow=
 
-; See qos.tex or Quality of Service section of asterisk.pdf for a description of these parameters.
+; See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for a description of these parameters.
 ;tos=cs3		; Sets TOS for signaling packets.
 ;tos_audio=ef		; Sets TOS for RTP audio packets.
 ;tos_video=af41		; Sets TOS for RTP video packets.

Modified: trunk/configs/sla.conf.sample
URL: http://svnview.digium.com/svn/asterisk/trunk/configs/sla.conf.sample?view=diff&rev=305843&r1=305842&r2=305843
==============================================================================
--- trunk/configs/sla.conf.sample (original)
+++ trunk/configs/sla.conf.sample Wed Feb  2 13:30:49 2011
@@ -21,11 +21,12 @@
 
 ;type=trunk                 ; This line is what marks this entry as a trunk.
 
-;device=DAHDI/3               ; Map this trunk declaration to a specific device.
+;device=DAHDI/3             ; Map this trunk declaration to a specific device.
                             ; NOTE: You can not just put any type of channel here.
                             ;       DAHDI channels can be directly used.  IP trunks
                             ;       require some indirect configuration which is
-                            ;       described in doc/asterisk.pdf.
+                            ;       described in 
+                            ; https://wiki.asterisk.org/wiki/display/AST/SLA+Trunk+Configuration
 
 ;autocontext=line1          ; This supports automatic generation of the dialplan entries
                             ; if the autocontext option is used.  Each trunk should have
@@ -61,8 +62,7 @@
 ;type=trunk
 ;device=Local/disa at line4_outbound ; A Local channel in combination with the Disa
                                   ; application can be used to support IP trunks.
-                                  ; See doc/asterisk.pdf on more information on how
-                                  ; IP trunks work.
+                                  ; See https://wiki.asterisk.org/wiki/display/AST/SLA+Trunk+Configuration
 ;autocontext=line4
 ; --------------------------------------
 

Modified: trunk/configs/unistim.conf.sample
URL: http://svnview.digium.com/svn/asterisk/trunk/configs/unistim.conf.sample?view=diff&rev=305843&r1=305842&r2=305843
==============================================================================
--- trunk/configs/unistim.conf.sample (original)
+++ trunk/configs/unistim.conf.sample Wed Feb  2 13:30:49 2011
@@ -5,7 +5,7 @@
 [general]
 port=5000                    ; UDP port
 ;
-; See qos.tex or Quality of Service section of asterisk.pdf for a description of these parameters.
+; See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for a description of these parameters.
 ;tos=cs3                ; Sets TOS for signaling packets.
 ;tos_audio=ef           ; Sets TOS for RTP audio packets.
 ;cos=3                  ; Sets 802.1p priority for signaling packets.

Modified: trunk/funcs/func_callcompletion.c
URL: http://svnview.digium.com/svn/asterisk/trunk/funcs/func_callcompletion.c?view=diff&rev=305843&r1=305842&r2=305843
==============================================================================
--- trunk/funcs/func_callcompletion.c (original)
+++ trunk/funcs/func_callcompletion.c Wed Feb  2 13:30:49 2011
@@ -58,7 +58,7 @@
 			a configuration parameter will only change the parameter for the
 			duration of the call.
 
-			For more information see <filename>doc/asterisk.pdf</filename>.
+			For more information see <filename>doc/AST.pdf</filename>.
 			For more information on call completion parameters, see <filename>configs/ccss.conf.sample</filename>.</para>
 		</description>
 	</function>

Modified: trunk/funcs/func_enum.c
URL: http://svnview.digium.com/svn/asterisk/trunk/funcs/func_enum.c?view=diff&rev=305843&r1=305842&r2=305843
==============================================================================
--- trunk/funcs/func_enum.c (original)
+++ trunk/funcs/func_enum.c Wed Feb  2 13:30:49 2011
@@ -127,7 +127,7 @@
 			</parameter>
 		</syntax>
 		<description>
-			<para>For more information see <filename>doc/asterisk.pdf</filename>.</para>
+			<para>For more information see <filename>doc/AST.pdf</filename>.</para>
 		</description>
 	</function>
 	<function name="TXTCIDNAME" language="en_US">




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