Replacing doc/* and asterisk.pdf with wiki links
Adding links to http(s)://wiki.asterisk.org git-svn-id: http://svn.digium.com/svn/asterisk/trunk@305843 f38db490-d61c-443f-a65b-d21fe96a405b
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@ -85,7 +85,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
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all DTMF events received on the channel, and notification if the channel is
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hung up. The received on the channel, and notification if the channel is hung
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up. The application will not be forcibly terminated when the channel is hung up.
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For more information see <filename>doc/asterisk.pdf</filename>.</para>
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For more information see <filename>doc/AST.pdf</filename>.</para>
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</description>
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</application>
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***/
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@ -27,7 +27,7 @@
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*
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* \par See also
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* \arg \ref Config_vm
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* \note For information about voicemail IMAP storage, read doc/asterisk.pdf
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* \note For information about voicemail IMAP storage, https://wiki.asterisk.org/wiki/display/AST/IMAP+Voicemail+Storage
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* \ingroup applications
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* \note This module requires res_adsi to load. This needs to be optional
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* during compilation.
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@ -27,7 +27,7 @@
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;bindaddr=0.0.0.0
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;port=4520
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;
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; See qos.tex or Quality of Service section of asterisk.pdf for a description of the tos parameter.
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; See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for a description of the tos parameter.
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;tos=ef
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;
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; Our entity identifier (Should generally be the MAC address of the
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@ -5,7 +5,7 @@
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port = 1720
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;bindaddr = 1.2.3.4 ; this SHALL contain a single, valid IP address for this machine
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;
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; See qos.tex or Quality of Service section of asterisk.pdf for a description of these parameters.
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; See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for a description of these parameters.
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;tos_audio=ef ; Sets TOS for RTP audio packets.
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;cos_audio=5 ; Sets 802.1p priority for RTP audio packets.
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;
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@ -259,7 +259,7 @@ forcejitterbuffer=no
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;
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;authdebug=no
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;
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; See qos.tex or Quality of Service section of asterisk.pdf for a description of these parameters.
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; See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for a description of these parameters.
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;tos=ef
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;cos=5
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;
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@ -53,7 +53,7 @@ codec=ulaw
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;
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flags=register,heartbeat
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;
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; See qos.tex or Quality of Service section of asterisk.pdf for a description of this parameter.
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; See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for a description of this parameter.
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;tos=ef
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;
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; Example iaxy provisioning
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@ -5,7 +5,7 @@
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;port = 2427
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;bindaddr = 0.0.0.0
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; See qos.tex or Quality of Service section of asterisk.pdf for a description of these parameters.
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; See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for a description of these parameters.
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;tos=cs3 ; Sets TOS for signaling packets.
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;tos_audio=ef ; Sets TOS for RTP audio packets.
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;cos=3 ; Sets 802.1p priority for signaling packets.
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@ -217,7 +217,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
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; and multiline formatted headers for strict
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; SIP compatibility (defaults to "yes")
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; See qos.tex or Quality of Service section of asterisk.pdf for a description of these parameters.
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; See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for a description of these parameters.
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;tos_sip=cs3 ; Sets TOS for SIP packets.
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;tos_audio=ef ; Sets TOS for RTP audio packets.
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;tos_video=af41 ; Sets TOS for RTP video packets.
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@ -29,7 +29,7 @@ keepalive=120
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; for framing options
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;disallow=
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; See qos.tex or Quality of Service section of asterisk.pdf for a description of these parameters.
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; See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for a description of these parameters.
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;tos=cs3 ; Sets TOS for signaling packets.
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;tos_audio=ef ; Sets TOS for RTP audio packets.
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;tos_video=af41 ; Sets TOS for RTP video packets.
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@ -21,11 +21,12 @@
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;type=trunk ; This line is what marks this entry as a trunk.
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;device=DAHDI/3 ; Map this trunk declaration to a specific device.
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;device=DAHDI/3 ; Map this trunk declaration to a specific device.
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; NOTE: You can not just put any type of channel here.
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; DAHDI channels can be directly used. IP trunks
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; require some indirect configuration which is
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; described in doc/asterisk.pdf.
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; described in
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; https://wiki.asterisk.org/wiki/display/AST/SLA+Trunk+Configuration
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;autocontext=line1 ; This supports automatic generation of the dialplan entries
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; if the autocontext option is used. Each trunk should have
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;type=trunk
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;device=Local/disa@line4_outbound ; A Local channel in combination with the Disa
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; application can be used to support IP trunks.
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; See doc/asterisk.pdf on more information on how
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; IP trunks work.
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; See https://wiki.asterisk.org/wiki/display/AST/SLA+Trunk+Configuration
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;autocontext=line4
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; --------------------------------------
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@ -5,7 +5,7 @@
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[general]
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port=5000 ; UDP port
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;
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; See qos.tex or Quality of Service section of asterisk.pdf for a description of these parameters.
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; See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for a description of these parameters.
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;tos=cs3 ; Sets TOS for signaling packets.
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;tos_audio=ef ; Sets TOS for RTP audio packets.
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;cos=3 ; Sets 802.1p priority for signaling packets.
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@ -58,7 +58,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
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a configuration parameter will only change the parameter for the
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duration of the call.
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For more information see <filename>doc/asterisk.pdf</filename>.
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For more information see <filename>doc/AST.pdf</filename>.
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For more information on call completion parameters, see <filename>configs/ccss.conf.sample</filename>.</para>
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</description>
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</function>
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@ -127,7 +127,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
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</parameter>
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</syntax>
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<description>
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<para>For more information see <filename>doc/asterisk.pdf</filename>.</para>
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<para>For more information see <filename>doc/AST.pdf</filename>.</para>
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</description>
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</function>
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<function name="TXTCIDNAME" language="en_US">
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