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asterisk
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Replacing doc/* and asterisk.pdf with wiki links

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



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@305843 65c4cc65-6c06-0410-ace0-fbb531ad65f3
10-digiumphones
Andrew Latham 15 years ago
parent 9f1a17f137
commit 93bade5639
13 changed files with 16 additions and 16 deletions
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2
apps/app_externalivr.c
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@ -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>
***/

2
apps/app_voicemail.c
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@ -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.

2
configs/dundi.conf.sample
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@ -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

2
configs/h323.conf.sample
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@ -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.
;

2
configs/iax.conf.sample
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@ -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
;

2
configs/iaxprov.conf.sample
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@ -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

2
configs/mgcp.conf.sample
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@ -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.

2
configs/sip.conf.sample
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@ -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.

2
configs/skinny.conf.sample
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@ -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.

6
configs/sla.conf.sample
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@ -25,7 +25,8 @@
; 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
; --------------------------------------

2
configs/unistim.conf.sample
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@ -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.

2
funcs/func_callcompletion.c
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@ -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>

2
funcs/func_enum.c
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@ -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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