Replacing doc/* with wiki links
Adding links to http(s)://wiki.asterisk.org git-svn-id: http://svn.digium.com/svn/asterisk/trunk@305799 f38db490-d61c-443f-a65b-d21fe96a405b
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@ -1,8 +1,8 @@
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;
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; --- Call Completion Supplementary Services ---
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;
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; For more information about CCSS, see the CCSS user documentation included
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; in the HTML and PDF documentation generated from the doc/tex/ directory.
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; For more information about CCSS, see the CCSS user documentation
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; https://wiki.asterisk.org/wiki/display/AST/Call+Completion+Supplementary+Services+(CCSS)
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;
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[general]
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@ -466,7 +466,7 @@ usecallerid=yes
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; polarity = polarity reversal signals the start
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; polarity_IN = polarity reversal signals the start, for India,
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; for dtmf dialtone detection; using DTMF.
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; (see doc/India-CID.txt)
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; (see https://wiki.asterisk.org/wiki/display/AST/Caller+ID+in+India)
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; dtmf = causes monitor loop to look for dtmf energy on the
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; incoming channel to initate cid acquisition
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;
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@ -1216,7 +1216,7 @@ pickupgroup=1
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;channel = 25-47
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;
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; For more information on setting up SS7, see the README file in libss7 or
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; the doc/ss7.txt file in the Asterisk source tree.
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; https://wiki.asterisk.org/wiki/display/AST/Signaling+System+Number+7
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; ----------------- SS7 Options ----------------------------------------
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; ---------------- Options for use with signalling=mfcr2 --------------
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@ -2,8 +2,8 @@
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; Static and realtime external configuration
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; engine configuration
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;
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; Please read doc/extconfig.txt for basic table
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; formatting information.
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; See https://wiki.asterisk.org/wiki/display/AST/Realtime+Database+Configuration
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; for basic table formatting information.
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;
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[settings]
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;
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@ -27,7 +27,8 @@ port = 1720
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;allow=all ; turns on all installed codecs
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;disallow=g723.1 ; Hm... Proprietary, don't use it...
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;allow=gsm ; Always allow GSM, it's cool :)
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;allow=ulaw ; see doc/rtp-packetization for framing options
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;allow=ulaw ; see https://wiki.asterisk.org/wiki/display/AST/RTP+Packetization
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; for framing options
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;
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; User-Input Mode (DTMF)
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;
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@ -10,7 +10,8 @@
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; If you use the subagent model, you need to enable agentx in snmpd.conf
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; Note that you can only run one Asterisk on the system in this case.
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;
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; Please read documentat in doc/snmp.txt to get more information about
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; See https://wiki.asterisk.org/wiki/display/AST/Simple+Network+Management+Protocol+(SNMP)+Support
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; to get more information about
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; snmp support in Asterisk
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[general]
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@ -274,7 +274,8 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
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;disallow=all ; First disallow all codecs
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;allow=ulaw ; Allow codecs in order of preference
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;allow=ilbc ; see doc/rtp-packetization for framing options
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;allow=ilbc ; see https://wiki.asterisk.org/wiki/display/AST/RTP+Packetization
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; for framing options
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;
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; This option specifies a preference for which music on hold class this channel
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; should listen to when put on hold if the music class has not been set on the
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@ -870,8 +871,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
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;----------------------------------------- REALTIME SUPPORT ------------------------
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; For additional information on ARA, the Asterisk Realtime Architecture,
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; please read realtime.txt and extconfig.txt in the /doc directory of the
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; source code.
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; please read https://wiki.asterisk.org/wiki/display/AST/Realtime+Database+Configuration
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;
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;rtcachefriends=yes ; Cache realtime friends by adding them to the internal list
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; just like friends added from the config file only on a
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@ -25,7 +25,8 @@ keepalive=120
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;
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;regcontext=skinnyregistrations
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;allow=all ; see doc/rtp-packetization for framing options
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;allow=all ; see https://wiki.asterisk.org/wiki/display/AST/RTP+Packetization
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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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@ -1,7 +1,7 @@
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;
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; Configuration for Shared Line Appearances (SLA).
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;
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; See doc/asterisk.pdf for more information.
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; See http://wiki.asterisk.org or doc/AST.pdf for more information.
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;
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; ---- General Options ----------------
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@ -468,7 +468,7 @@ int ast_yyerror (const char *s, yyltype *loc, struct parse_io *parseio )
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(extra_error_message_supplied?extra_error_message:""), s2, parseio->string,spacebuf2);
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#endif
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#ifndef STANDALONE
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ast_log(LOG_WARNING,"If you have questions, please refer to doc/tex/channelvariables.tex.\n");
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ast_log(LOG_WARNING,"If you have questions, please refer to https://wiki.asterisk.org/wiki/display/AST/Channel+Variables\n");
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#endif
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free(s2);
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return(0);
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@ -2603,7 +2603,7 @@ int ast_yyerror (const char *s, yyltype *loc, struct parse_io *parseio )
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(extra_error_message_supplied?extra_error_message:""), s2, parseio->string,spacebuf2);
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#endif
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#ifndef STANDALONE
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ast_log(LOG_WARNING,"If you have questions, please refer to doc/tex/channelvariables.tex.\n");
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ast_log(LOG_WARNING,"If you have questions, please refer to https://wiki.asterisk.org/wiki/display/AST/Channel+Variables\n");
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#endif
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free(s2);
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return(0);
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@ -23,7 +23,7 @@
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* \author Mark Spencer <markster@digium.com>
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*
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* Includes the Asterisk Realtime API - ARA
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* See doc/realtime.txt and doc/extconfig.txt
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* See http://wiki.asterisk.org
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*/
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#include "asterisk.h"
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@ -32,7 +32,7 @@
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<depend>srtp</depend>
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***/
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/* See doc/tex/secure-calls.tex for SRTP usage information */
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/* See https://wiki.asterisk.org/wiki/display/AST/Secure+Calling */
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#include "asterisk.h"
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@ -137,7 +137,7 @@ static unsigned int dahdi_timer_get_max_rate(int handle)
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return 1000;
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}
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#define SEE_TIMING "For more information on Asterisk timing modules, including ways to potentially fix this problem, please see doc/timing.txt\n"
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#define SEE_TIMING "For more information on Asterisk timing modules, including ways to potentially fix this problem, please see https://wiki.asterisk.org/wiki/display/AST/Timing+Interfaces\n"
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static int dahdi_test_timer(void)
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{
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