From 93bade5639321193ae16a0ca656c701f45e8fbb5 Mon Sep 17 00:00:00 2001
From: Andrew Latham <lathama@gmail.com>
Date: Wed, 2 Feb 2011 19:30:49 +0000
Subject: [PATCH] Replacing doc/* and asterisk.pdf with wiki links

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



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@305843 65c4cc65-6c06-0410-ace0-fbb531ad65f3
---
 apps/app_externalivr.c      | 2 +-
 apps/app_voicemail.c        | 2 +-
 configs/dundi.conf.sample   | 2 +-
 configs/h323.conf.sample    | 2 +-
 configs/iax.conf.sample     | 2 +-
 configs/iaxprov.conf.sample | 2 +-
 configs/mgcp.conf.sample    | 2 +-
 configs/sip.conf.sample     | 2 +-
 configs/skinny.conf.sample  | 2 +-
 configs/sla.conf.sample     | 8 ++++----
 configs/unistim.conf.sample | 2 +-
 funcs/func_callcompletion.c | 2 +-
 funcs/func_enum.c           | 2 +-
 13 files changed, 16 insertions(+), 16 deletions(-)

diff --git a/apps/app_externalivr.c b/apps/app_externalivr.c
index 093e9ae272..0856ffe671 100644
--- a/apps/app_externalivr.c
+++ b/apps/app_externalivr.c
@@ -85,7 +85,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 			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>
  ***/
diff --git a/apps/app_voicemail.c b/apps/app_voicemail.c
index 2d593f379b..166df4c94a 100644
--- a/apps/app_voicemail.c
+++ b/apps/app_voicemail.c
@@ -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.
diff --git a/configs/dundi.conf.sample b/configs/dundi.conf.sample
index a3713412ec..4733733caf 100644
--- a/configs/dundi.conf.sample
+++ b/configs/dundi.conf.sample
@@ -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
diff --git a/configs/h323.conf.sample b/configs/h323.conf.sample
index 00eb73f0b2..80ec4830cc 100644
--- a/configs/h323.conf.sample
+++ b/configs/h323.conf.sample
@@ -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.
 ;
diff --git a/configs/iax.conf.sample b/configs/iax.conf.sample
index 73ae78afbc..a3d0fea127 100644
--- a/configs/iax.conf.sample
+++ b/configs/iax.conf.sample
@@ -259,7 +259,7 @@ forcejitterbuffer=no
 ;
 ;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
 ;
diff --git a/configs/iaxprov.conf.sample b/configs/iaxprov.conf.sample
index d3789dcdec..668809c4db 100644
--- a/configs/iaxprov.conf.sample
+++ b/configs/iaxprov.conf.sample
@@ -53,7 +53,7 @@ codec=ulaw
 ;
 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
diff --git a/configs/mgcp.conf.sample b/configs/mgcp.conf.sample
index f5e5473d1b..7c725bc3d4 100644
--- a/configs/mgcp.conf.sample
+++ b/configs/mgcp.conf.sample
@@ -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.
diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample
index 9507c71df9..9127628822 100644
--- a/configs/sip.conf.sample
+++ b/configs/sip.conf.sample
@@ -217,7 +217,7 @@ srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
                                 ; 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.
diff --git a/configs/skinny.conf.sample b/configs/skinny.conf.sample
index 163f557155..2199af19db 100644
--- a/configs/skinny.conf.sample
+++ b/configs/skinny.conf.sample
@@ -29,7 +29,7 @@ keepalive=120
 			; 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.
diff --git a/configs/sla.conf.sample b/configs/sla.conf.sample
index cf760af14a..5027009796 100644
--- a/configs/sla.conf.sample
+++ b/configs/sla.conf.sample
@@ -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@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
 ; --------------------------------------
 
diff --git a/configs/unistim.conf.sample b/configs/unistim.conf.sample
index d3f781ded8..52dc65e38b 100644
--- a/configs/unistim.conf.sample
+++ b/configs/unistim.conf.sample
@@ -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.
diff --git a/funcs/func_callcompletion.c b/funcs/func_callcompletion.c
index 6a7ced201f..b3fab310f4 100644
--- a/funcs/func_callcompletion.c
+++ b/funcs/func_callcompletion.c
@@ -58,7 +58,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 			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>
diff --git a/funcs/func_enum.c b/funcs/func_enum.c
index bdda66d2f7..378187ec39 100644
--- a/funcs/func_enum.c
+++ b/funcs/func_enum.c
@@ -127,7 +127,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 			</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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