diff --git a/BUGS b/BUGS
index efb9bbd796..cb9825b385 100644
--- a/BUGS
+++ b/BUGS
@@ -10,7 +10,7 @@ For more information on using the bug tracker, or to
learn how you can contribute by acting as a bug marshal
please see:
- https://wiki.asterisk.org/wiki/x/RgAtAQ
+ https://docs.asterisk.org/Asterisk-Community/Asterisk-Issue-Guidelines/
If you would like to submit a feature request, please
resist the temptation to post it to the bug tracker.
diff --git a/README-SERIOUSLY.bestpractices.md b/README-SERIOUSLY.bestpractices.md
index fa4b0a6fa2..192bdaae26 100644
--- a/README-SERIOUSLY.bestpractices.md
+++ b/README-SERIOUSLY.bestpractices.md
@@ -379,9 +379,8 @@ is set to no.
In Asterisk 12 and later, live_dangerously defaults to no.
-
-[voip-security-webinar]: https://www.asterisk.org/security/webinar/
-[blog-sip-security]: http://blogs.digium.com/2009/03/28/sip-security/
+[voip-security-webinar]: https://docs.asterisk.org/Deployment/Important-Security-Considerations/Asterisk-Security-Webinars/
+[blog-sip-security]: https://web.archive.org/web/20171030134647/http://blogs.digium.com/2009/03/28/sip-security/
[Strong Password Generator]: https://www.strongpasswordgenerator.com
[Filtering Data]: #filtering-data
[Proper Device Naming]: #proper-device-naming
@@ -389,4 +388,4 @@ In Asterisk 12 and later, live_dangerously defaults to no.
[Reducing Pattern Match Typos]: #reducing-pattern-match-typos
[Manager Class Authorizations]: #manager-class-authorizations
[Avoid Privilege Escalations]: #avoid-privilege-escalations
-[Important Security Considerations]: https://wiki.asterisk.org/wiki/display/AST/Important+Security+Considerations
+[Important Security Considerations]: https://docs.asterisk.org/Deployment/Important-Security-Considerations/
diff --git a/README.md b/README.md
index 0eb4b879a5..feb138c211 100644
--- a/README.md
+++ b/README.md
@@ -20,7 +20,7 @@ more telephony interfaces than just Internet telephony. Asterisk also has a
vast amount of support for traditional PSTN telephony, as well.
For more information on the project itself, please visit the Asterisk
-[home page] and the official [wiki]. In addition you'll find lots
+[home page] and the official [documentation]. In addition you'll find lots
of information compiled by the Asterisk community at [voip-info.org].
There is a book on Asterisk published by O'Reilly under the Creative Commons
@@ -258,7 +258,7 @@ Asterisk is a trademark of Sangoma Technologies Corporation
[home page]: https://www.asterisk.org
[support]: https://www.asterisk.org/support
-[wiki]: https://wiki.asterisk.org/
+[documentation]: https://docs.asterisk.org/
[mailing list]: http://lists.digium.com/mailman/listinfo/asterisk-users
[chan_dahdi.conf]: configs/samples/chan_dahdi.conf.sample
[voip-info.org]: http://www.voip-info.org/wiki-Asterisk
@@ -269,4 +269,4 @@ Asterisk is a trademark of Sangoma Technologies Corporation
[CHANGES]: CHANGES
[configs]: configs
[doc]: doc
-[Important Security Considerations]: https://wiki.asterisk.org/wiki/display/AST/Important+Security+Considerations
+[Important Security Considerations]: https://docs.asterisk.org/Deployment/Important-Security-Considerations/
diff --git a/apps/app_audiosocket.c b/apps/app_audiosocket.c
index ded88ccf7f..7270937d84 100644
--- a/apps/app_audiosocket.c
+++ b/apps/app_audiosocket.c
@@ -61,7 +61,7 @@
Connects to the given TCP service, then transmits channel audio over that socket. In turn, audio is received from the socket and sent to the channel. Only audio frames will be transmitted.
- Protocol is specified at https://wiki.asterisk.org/wiki/display/AST/AudioSocket
+ Protocol is specified at https://docs.asterisk.org/Configuration/Channel-Drivers/AudioSocket/
This application does not automatically answer and should generally be preceeded by an application such as Answer() or Progress().
diff --git a/apps/app_skel.c b/apps/app_skel.c
index b59ebe56bd..01bc8f98cc 100644
--- a/apps/app_skel.c
+++ b/apps/app_skel.c
@@ -16,7 +16,7 @@
* at the top of the source tree.
*
* Please follow coding guidelines
- * https://wiki.asterisk.org/wiki/display/AST/Coding+Guidelines
+ * https://docs.asterisk.org/Development/Policies-and-Procedures/Coding-Guidelines/
*/
/*! \file
diff --git a/apps/app_voicemail.c b/apps/app_voicemail.c
index e4067abd06..2ffdd7674d 100644
--- a/apps/app_voicemail.c
+++ b/apps/app_voicemail.c
@@ -27,7 +27,7 @@
*
* \par See also
* \arg \ref voicemail.conf "Config_voicemail"
- * \note For information about voicemail IMAP storage, https://wiki.asterisk.org/wiki/display/AST/IMAP+Voicemail+Storage
+ * \note For information about voicemail IMAP storage, https://docs.asterisk.org/Configuration/Applications/Voicemail/IMAP-Voicemail-Storage/
* \ingroup applications
* \todo This module requires res_adsi to load. This needs to be optional
* during compilation.
diff --git a/apps/confbridge/include/conf_state.h b/apps/confbridge/include/conf_state.h
index a9760c9012..3cd0c5b8bd 100644
--- a/apps/confbridge/include/conf_state.h
+++ b/apps/confbridge/include/conf_state.h
@@ -22,7 +22,7 @@
*
* \author\verbatim Terry Wilson \endverbatim
*
- * See https://wiki.asterisk.org/wiki/display/AST/Confbridge+state+changes for
+ * See https://docs.asterisk.org/Development/Reference-Information/Other-Reference-Information/Confbridge-state-changes/ for
* a more complete description of how conference states work.
*/
diff --git a/configs/basic-pbx/README b/configs/basic-pbx/README
index 0f57ad6c25..c5c19554ee 100644
--- a/configs/basic-pbx/README
+++ b/configs/basic-pbx/README
@@ -8,8 +8,8 @@ If you intend to use this configuration as a template for your own, then
you will need to change many values in the various configuration files to
match your own devices, network, SIP ITSP accounts and more.
-For further documentation on this configuration see the Asterisk wiki:
-https://wiki.asterisk.org/wiki/display/AST/Reference+Use+Cases+for+Asterisk.
+For further documentation on this configuration see the Asterisk documentation:
+https://docs.asterisk.org/Deployment/Reference-Use-Cases-for-Asterisk/.
Please report bugs or errors in configuration on the Asterisk issue tracker:
-https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines
+https://docs.asterisk.org/Asterisk-Community/Asterisk-Issue-Guidelines/
diff --git a/configs/samples/ccss.conf.sample b/configs/samples/ccss.conf.sample
index 031b18b7a4..efb24928ae 100644
--- a/configs/samples/ccss.conf.sample
+++ b/configs/samples/ccss.conf.sample
@@ -2,7 +2,7 @@
; --- Call Completion Supplementary Services ---
;
; For more information about CCSS, see the CCSS user documentation
-; https://wiki.asterisk.org/wiki/display/AST/Call+Completion+Supplementary+Services+(CCSS)
+; https://docs.asterisk.org/Deployment/PSTN-Connectivity/Call-Completion-Supplementary-Services-CCSS/
;
[general]
diff --git a/configs/samples/chan_dahdi.conf.sample b/configs/samples/chan_dahdi.conf.sample
index 3979a7648f..4640407924 100644
--- a/configs/samples/chan_dahdi.conf.sample
+++ b/configs/samples/chan_dahdi.conf.sample
@@ -595,7 +595,7 @@ usecallerid=yes
; polarity = polarity reversal signals the start
; polarity_IN = polarity reversal signals the start, for India,
; for dtmf dialtone detection; using DTMF.
-; (see https://wiki.asterisk.org/wiki/display/AST/Caller+ID+in+India)
+; (see https://wiki.asterisk.org/wiki/display/AST/Caller+ID+in+India)
; dtmf = causes monitor loop to look for dtmf energy on the
; incoming channel to initate cid acquisition
;
@@ -1579,7 +1579,7 @@ pickupgroup=1
;#include ss7.timers
; For more information on setting up SS7, see the README file in libss7 or
-; https://wiki.asterisk.org/wiki/display/AST/Signaling+System+Number+7
+; https://docs.asterisk.org/Deployment/PSTN-Connectivity/Signaling-System-Number-7/
; ----------------- SS7 Options ----------------------------------------
; ---------------- Options for use with signalling=mfcr2 --------------
diff --git a/configs/samples/extconfig.conf.sample b/configs/samples/extconfig.conf.sample
index f5de687325..df154381d2 100644
--- a/configs/samples/extconfig.conf.sample
+++ b/configs/samples/extconfig.conf.sample
@@ -2,7 +2,7 @@
; Static and realtime external configuration
; engine configuration
;
-; See https://wiki.asterisk.org/wiki/display/AST/Realtime+Database+Configuration
+; See https://docs.asterisk.org/Fundamentals/Asterisk-Configuration/Database-Support-Configuration/Realtime-Database-Configuration/
; for basic table formatting information.
;
[settings]
diff --git a/configs/samples/geolocation.conf.sample b/configs/samples/geolocation.conf.sample
index fdb9614b9c..9348ae5542 100644
--- a/configs/samples/geolocation.conf.sample
+++ b/configs/samples/geolocation.conf.sample
@@ -1,7 +1,7 @@
;--
Geolocation Profile Sample Configuration
- Please see https://wiki.asterisk.org/wiki/display/AST/Geolocation
+ Please see https://docs.asterisk.org/Deployment/Geolocation/
for the most current information.
--;
@@ -33,7 +33,7 @@ incoming calls (Asterisk is the UAS) and and one for outgoing calls
NOTE:
-See https://wiki.asterisk.org/wiki/display/AST/Geolocation for the most
+See https://docs.asterisk.org/Deployment/Geolocation/ for the most
complete and up-to-date information on valid values for the object
parameters and a full list of references.
@@ -96,7 +96,7 @@ variables like ${EXTEN}, channel variables you may have added in the
dialplan, or variables you may have specified in the profile that
references this location object.
-NOTE: See https://wiki.asterisk.org/wiki/display/AST/Geolocation for the
+NOTE: See https://docs.asterisk.org/Deployment/Geolocation/ for the
most complete and up-to-date information on valid values for the object
parameters and a full list of references.
diff --git a/configs/samples/pjsip.conf.sample b/configs/samples/pjsip.conf.sample
index e31f4d5a06..247b540276 100644
--- a/configs/samples/pjsip.conf.sample
+++ b/configs/samples/pjsip.conf.sample
@@ -20,7 +20,7 @@
; Documentation
;
-; The official documentation is at http://wiki.asterisk.org
+; The official documentation is at https://docs.asterisk.org
; You can read the XML configuration help via Asterisk command line with
; "config show help res_pjsip", then you can drill down through the various
; sections and their options.
@@ -31,8 +31,8 @@
; At a minimum please read the file "README-SERIOUSLY.bestpractices.txt",
; located in the Asterisk source directory before starting Asterisk.
; Otherwise you risk allowing the security of the Asterisk system to be
-; compromised. Beyond that please visit and read the security information on
-; the wiki at: https://wiki.asterisk.org/wiki/x/EwFB
+; compromised. Beyond that please visit and read the security information in
+; the documentation at: https://docs.asterisk.org/Deployment/Important-Security-Considerations/
;
; A few basics to pay attention to:
;
@@ -47,7 +47,7 @@
;
; See the example ACL configuration in this file. Read the configuration help
; for the section and all of its options. Look over the samples in acl.conf
-; and documentation at https://wiki.asterisk.org/wiki/x/uA80AQ
+; and documentation at https://docs.asterisk.org/Configuration/Core-Configuration/Named-ACLs/
; If possible, restrict access to only networks and addresses you trust.
;
; Dialplan Contexts
@@ -393,7 +393,7 @@
;rewrite_contact=yes ; necessary if endpoint does not know/register public ip:port
;ice_support=yes ;This is specific to clients that support NAT traversal
;for media via ICE,STUN,TURN. See the wiki at:
- ;https://wiki.asterisk.org/wiki/x/D4FHAQ
+ ;https://docs.asterisk.org/Configuration/Miscellaneous/Interactive-Connectivity-Establishment-ICE-in-Asterisk/
;for a deeper explanation of this topic.
;[6002]
@@ -1454,7 +1454,7 @@
; MODULE PROVIDING BELOW SECTION(S): res_pjsip_outbound_publish
;======================OUTBOUND_PUBLISH SECTION OPTIONS=====================
-; See https://wiki.asterisk.org/wiki/display/AST/Publishing+Extension+State
+; See https://docs.asterisk.org/Configuration/Channel-Drivers/SIP/Configuring-res_pjsip/Publishing-Extension-State/
; for more information.
;[outbound-publish]
;type=outbound-publish ; Must be of type 'outbound-publish'.
@@ -1509,9 +1509,9 @@
; MODULE PROVIDING BELOW SECTION(S): res_pjsip_pubsub
-;=============================RESOURCE-LIST===================================
-; See https://wiki.asterisk.org/wiki/pages/viewpage.action?pageId=30278158
+; See https://docs.asterisk.org/Configuration/Channel-Drivers/SIP/Configuring-res_pjsip/Resource-List-Subscriptions-RLS/
; for more information.
+;=============================RESOURCE-LIST===================================
;[resource_list]
;type=resource_list ; Must be of type 'resource_list'.
@@ -1568,7 +1568,7 @@
;==========================INBOUND_PUBLICATION================================
-; See https://wiki.asterisk.org/wiki/display/AST/Exchanging+Device+and+Mailbox+State+Using+PJSIP
+; See https://docs.asterisk.org/Configuration/Channel-Drivers/SIP/Configuring-res_pjsip/Exchanging-Device-and-Mailbox-State-Using-PJSIP/
; for more information.
;[inbound-publication]
;type= ; Must be of type 'inbound-publication'.
@@ -1579,7 +1579,7 @@
; MODULE PROVIDING BELOW SECTION(S): res_pjsip_publish_asterisk
;==========================ASTERISK_PUBLICATION===============================
-; See https://wiki.asterisk.org/wiki/display/AST/Exchanging+Device+and+Mailbox+State+Using+PJSIP
+; See https://docs.asterisk.org/Configuration/Channel-Drivers/SIP/Configuring-res_pjsip/Exchanging-Device-and-Mailbox-State-Using-PJSIP/
; for more information.
;[asterisk-publication]
;type=asterisk-publication ; Must be of type 'asterisk-publication'.
diff --git a/configs/samples/pjsip_wizard.conf.sample b/configs/samples/pjsip_wizard.conf.sample
index 5de28b3046..97e0c6da5e 100644
--- a/configs/samples/pjsip_wizard.conf.sample
+++ b/configs/samples/pjsip_wizard.conf.sample
@@ -20,7 +20,7 @@
; Documentation
;
-; The official documentation is at http://wiki.asterisk.org
+; The official documentation is at https://docs.asterisk.org
; You can read the XML configuration help via Asterisk command line with
; "config show help res_pjsip_config_wizard", then you can drill down through
; the various sections and their options.
diff --git a/configs/samples/sla.conf.sample b/configs/samples/sla.conf.sample
index 1f5a56e7bf..70da88ae71 100644
--- a/configs/samples/sla.conf.sample
+++ b/configs/samples/sla.conf.sample
@@ -1,7 +1,7 @@
;
; Configuration for Shared Line Appearances (SLA).
;
-; See http://wiki.asterisk.org or doc/AST.pdf for more information.
+; See https://docs.asterisk.org for more information.
;
; ---- General Options ----------------
@@ -37,7 +37,7 @@
; DAHDI channels can be directly used. IP trunks
; require some indirect configuration which is
; described in
- ; https://wiki.asterisk.org/wiki/display/AST/SLA+Trunk+Configuration
+ ; https://docs.asterisk.org/Configuration/Applications/Shared-Line-Appearances-SLA/
;autocontext=line1 ; This supports automatic generation of the dialplan entries
; if the autocontext option is used. Each trunk should have
@@ -73,7 +73,7 @@
;type=trunk
;device=Local/disa@line4_outbound ; A Local channel in combination with the Disa
; application can be used to support IP trunks.
- ; See https://wiki.asterisk.org/wiki/display/AST/SLA+Trunk+Configuration
+ ; See https://docs.asterisk.org/Configuration/Applications/Shared-Line-Appearances-SLA/
;autocontext=line4
; --------------------------------------
diff --git a/configs/samples/stir_shaken.conf.sample b/configs/samples/stir_shaken.conf.sample
index 677d3bb3ba..bc4220e8fc 100644
--- a/configs/samples/stir_shaken.conf.sample
+++ b/configs/samples/stir_shaken.conf.sample
@@ -24,7 +24,7 @@
; config directory is.
;
; Visit the wiki page:
-; https://wiki.asterisk.org/wiki/display/AST/STIR+and+SHAKEN
+; https://docs.asterisk.org/Deployment/STIR-SHAKEN/
;
; [general]
;
diff --git a/doc/CODING-GUIDELINES b/doc/CODING-GUIDELINES
index 8029d4d68d..ce7807979a 100644
--- a/doc/CODING-GUIDELINES
+++ b/doc/CODING-GUIDELINES
@@ -1,2 +1,2 @@
Coding guidelines are available on the Asterisk wiki at:
-https://wiki.asterisk.org/wiki/display/AST/Coding+Guidelines
+https://docs.asterisk.org/Development/Policies-and-Procedures/Coding-Guidelines/
diff --git a/doc/README.txt b/doc/README.txt
index e9b935cf0c..f4e6134981 100644
--- a/doc/README.txt
+++ b/doc/README.txt
@@ -1,13 +1,7 @@
The vast majority of the Asterisk project documentation has been moved to the
-project wiki:
+project documentation:
- https://wiki.asterisk.org/
-
-Asterisk release tarballs contain an export of the wiki in PDF and plain text
-form, which you can find in:
-
- doc/AST.pdf
- doc/AST.txt
+ https://docs.asterisk.org/
Asterisk uses the Doxygen documentation software. Run "make progdocs" and open
the resulting documentation index at doc/api/index.html in a webbrowser or copy
diff --git a/doc/asterisk.8 b/doc/asterisk.8
index c3bc6d1546..bc6f36969d 100644
--- a/doc/asterisk.8
+++ b/doc/asterisk.8
@@ -248,7 +248,7 @@ https://www.asterisk.org - The Asterisk Home Page
.PP
http://www.asteriskdocs.org - The Asterisk Documentation Project
.PP
-https://wiki.asterisk.org - The Asterisk Wiki
+https://docs.asterisk.org - The Asterisk documentation
.PP
https://www.digium.com/ - Asterisk is sponsored by Digium
.SH AUTHOR
diff --git a/doc/asterisk.sgml b/doc/asterisk.sgml
index 3620b71ff7..c13b0ba4d0 100644
--- a/doc/asterisk.sgml
+++ b/doc/asterisk.sgml
@@ -427,7 +427,7 @@
http://www.asteriskdocs.org - The Asterisk Documentation Project
- https://wiki.asterisk.org - The Asterisk Wiki
+ https://docs.asterisk.org/ - The Asterisk documentation
https://www.digium.com/ - Asterisk is sponsored by Digium
diff --git a/doc/lang/language-criteria.txt b/doc/lang/language-criteria.txt
index 30a09cb58b..b80fb88e88 100644
--- a/doc/lang/language-criteria.txt
+++ b/doc/lang/language-criteria.txt
@@ -1,3 +1,3 @@
This document has been moved to the Asterisk Wiki:
-https://wiki.asterisk.org/wiki/display/AST/Asterisk+Sounds+Submission+Process
+https://docs.asterisk.org/Development/Policies-and-Procedures/Asterisk-Sounds-Submission-Process/
diff --git a/main/ast_expr2.fl b/main/ast_expr2.fl
index 542f01817f..d0b2f8c671 100644
--- a/main/ast_expr2.fl
+++ b/main/ast_expr2.fl
@@ -468,7 +468,7 @@ int ast_yyerror (const char *s, yyltype *loc, struct parse_io *parseio )
(extra_error_message_supplied ? extra_error_message : ""), s2, parseio->string, spacebuf);
#endif
#ifndef STANDALONE
- ast_log(LOG_WARNING,"If you have questions, please refer to https://wiki.asterisk.org/wiki/display/AST/Channel+Variables\n");
+ ast_log(LOG_WARNING,"If you have questions, please refer to https://docs.asterisk.org/Configuration/Dialplan/Variables/Channel-Variables/\n");
#endif
free(s2);
return(0);
diff --git a/main/ast_expr2f.c b/main/ast_expr2f.c
index 9819eb7c5a..9144a084ec 100644
--- a/main/ast_expr2f.c
+++ b/main/ast_expr2f.c
@@ -2604,7 +2604,7 @@ int ast_yyerror (const char *s, yyltype *loc, struct parse_io *parseio )
(extra_error_message_supplied ? extra_error_message : ""), s2, parseio->string, spacebuf);
#endif
#ifndef STANDALONE
- ast_log(LOG_WARNING,"If you have questions, please refer to https://wiki.asterisk.org/wiki/display/AST/Channel+Variables\n");
+ ast_log(LOG_WARNING,"If you have questions, please refer to https://docs.asterisk.org/Configuration/Dialplan/Variables/Channel-Variables/\n");
#endif
free(s2);
return(0);
diff --git a/main/asterisk.c b/main/asterisk.c
index 650591d6ff..0d306fdeeb 100644
--- a/main/asterisk.c
+++ b/main/asterisk.c
@@ -70,8 +70,8 @@
/*!
* \page asterisk_community_resources Asterisk Community Resources
* \par Websites
- * \li http://www.asterisk.org Asterisk Homepage
- * \li http://wiki.asterisk.org Asterisk Wiki
+ * \li https://www.asterisk.org Asterisk Homepage
+ * \li https://docs.asterisk.org Asterisk documentation
*
* \par Mailing Lists
* \par
diff --git a/main/config.c b/main/config.c
index 3dc47088de..dd8dacfcb8 100644
--- a/main/config.c
+++ b/main/config.c
@@ -23,7 +23,7 @@
* \author Mark Spencer
*
* Includes the Asterisk Realtime API - ARA
- * See http://wiki.asterisk.org
+ * See https://docs.asterisk.org
*/
/*** MODULEINFO
diff --git a/main/pbx_functions.c b/main/pbx_functions.c
index 081c33f6ed..fd542c9568 100644
--- a/main/pbx_functions.c
+++ b/main/pbx_functions.c
@@ -467,7 +467,7 @@ void pbx_live_dangerously(int new_live_dangerously)
{
if (new_live_dangerously && !live_dangerously) {
ast_log(LOG_WARNING, "Privilege escalation protection disabled!\n"
- "See https://wiki.asterisk.org/wiki/x/1gKfAQ for more details.\n");
+ "See https://docs.asterisk.org/Configuration/Dialplan/Privilege-Escalations-with-Dialplan-Functions/ for more details.\n");
}
if (!new_live_dangerously && live_dangerously) {
diff --git a/main/stasis.c b/main/stasis.c
index 0715e4adde..05a7a505f7 100644
--- a/main/stasis.c
+++ b/main/stasis.c
@@ -286,8 +286,8 @@
* \par Subscriber shutdown sequencing
*
* Subscribers are sensitive to shutdown sequencing, specifically in how the
- * reference message types. This is fully detailed on the wiki at
- * https://wiki.asterisk.org/wiki/x/K4BqAQ.
+ * reference message types. This is fully detailed in the documentation at
+ * https://docs.asterisk.org/Development/Roadmap/Asterisk-12-Projects/Asterisk-12-API-Improvements/Stasis-Message-Bus/Using-the-Stasis-Message-Bus/Stasis-Subscriber-Shutdown-Problem/.
*
* In short, the lifetime of the \a data (and \a callback, if in a module) must
* be held until the stasis_subscription_final_message() has been received.
diff --git a/res/ari/resource_channels.h b/res/ari/resource_channels.h
index a16d9be31b..4110301a6e 100644
--- a/res/ari/resource_channels.h
+++ b/res/ari/resource_channels.h
@@ -209,7 +209,7 @@ void ast_ari_channels_originate_with_id(struct ast_variable *headers, struct ast
struct ast_ari_channels_hangup_args {
/*! Channel's id */
const char *channel_id;
- /*! The reason code for hanging up the channel for detail use. Mutually exclusive with 'reason'. See detail hangup codes at here. https://wiki.asterisk.org/wiki/display/AST/Hangup+Cause+Mappings */
+ /*! The reason code for hanging up the channel for detail use. Mutually exclusive with 'reason'. See detail hangup codes at here. https://docs.asterisk.org/Configuration/Miscellaneous/Hangup-Cause-Mappings/ */
const char *reason_code;
/*! Reason for hanging up the channel for simple use. Mutually exclusive with 'reason_code'. */
const char *reason;
diff --git a/res/res_ari.c b/res/res_ari.c
index 025fa90ca4..e094f24d98 100644
--- a/res/res_ari.c
+++ b/res/res_ari.c
@@ -93,7 +93,7 @@
[http.conf]
- [https://wiki.asterisk.org/wiki/display/AST/Asterisk+Builtin+mini-HTTP+Server]
+ [https://docs.asterisk.org/Configuration/Core-Configuration/Asterisk-Builtin-mini-HTTP-Server/]
diff --git a/res/res_pjsip/pjsip_config.xml b/res/res_pjsip/pjsip_config.xml
index 5c64f0282d..a7ccf741c8 100644
--- a/res/res_pjsip/pjsip_config.xml
+++ b/res/res_pjsip/pjsip_config.xml
@@ -385,8 +385,8 @@
setup time.
- A more detailed description of how this option functions can be found on
- the Asterisk wiki https://wiki.asterisk.org/wiki/display/AST/SIP+Direct+Media+Reinvite+Glare+Avoidance
+ A more detailed description of how this option functions can be found in
+ the Asterisk documentation https://docs.asterisk.org/Configuration/Channel-Drivers/SIP/Concepts/SIP-Direct-Media-Reinvite-Glare-Avoidance/
diff --git a/res/res_pjsip_config_wizard.c b/res/res_pjsip_config_wizard.c
index 59976b1584..91228da05c 100644
--- a/res/res_pjsip_config_wizard.c
+++ b/res/res_pjsip_config_wizard.c
@@ -111,7 +111,7 @@
For more information, visit:
- https://wiki.asterisk.org/wiki/display/AST/PJSIP+Configuration+Wizard
+ https://docs.asterisk.org/Configuration/Channel-Drivers/SIP/Configuring-res_pjsip/PJSIP-Configuration-Wizard/
@@ -119,7 +119,7 @@
Provides config wizard.
For more information, visit:
- https://wiki.asterisk.org/wiki/display/AST/PJSIP+Configuration+Wizard
+ https://docs.asterisk.org/Configuration/Channel-Drivers/SIP/Configuring-res_pjsip/PJSIP-Configuration-Wizard/
Must be 'wizard'.
@@ -214,7 +214,7 @@
Normal dialplan precedence rules apply so if there's already a hint for
this extension in hint_context, this one will be ignored.
For more information, visit:
- https://wiki.asterisk.org/wiki/display/AST/PJSIP+Configuration+Wizard
+ https://docs.asterisk.org/Configuration/Channel-Drivers/SIP/Configuring-res_pjsip/PJSIP-Configuration-Wizard/
@@ -235,7 +235,7 @@
Normal dialplan precedence rules apply so if there's already a priority 1
application for this specific extension in hint_context,
this one will be ignored. For more information, visit:
- https://wiki.asterisk.org/wiki/display/AST/PJSIP+Configuration+Wizard
+ https://docs.asterisk.org/Configuration/Channel-Drivers/SIP/Configuring-res_pjsip/PJSIP-Configuration-Wizard/
diff --git a/res/res_srtp.c b/res/res_srtp.c
index e10421cbb4..33786d020a 100644
--- a/res/res_srtp.c
+++ b/res/res_srtp.c
@@ -35,7 +35,7 @@
core
***/
-/* See https://wiki.asterisk.org/wiki/display/AST/Secure+Calling */
+/* See https://docs.asterisk.org/Deployment/Secure-Calling/ */
#include "asterisk.h" /* for NULL, size_t, memcpy, etc */
diff --git a/res/res_timing_dahdi.c b/res/res_timing_dahdi.c
index c49f057ac9..2b3d885cee 100644
--- a/res/res_timing_dahdi.c
+++ b/res/res_timing_dahdi.c
@@ -170,7 +170,7 @@ static int dahdi_timer_fd(void *data)
return timer->fd;
}
-#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"
+#define SEE_TIMING "For more information on Asterisk timing modules, including ways to potentially fix this problem, please see https://docs.asterisk.org/Configuration/Core-Configuration/Timing-Interfaces/\n"
static int dahdi_test_timer(void)
{
diff --git a/rest-api/api-docs/channels.json b/rest-api/api-docs/channels.json
index 9065803968..3f8e173a62 100644
--- a/rest-api/api-docs/channels.json
+++ b/rest-api/api-docs/channels.json
@@ -416,7 +416,7 @@
},
{
"name": "reason_code",
- "description": "The reason code for hanging up the channel for detail use. Mutually exclusive with 'reason'. See detail hangup codes at here. https://wiki.asterisk.org/wiki/display/AST/Hangup+Cause+Mappings",
+ "description": "The reason code for hanging up the channel for detail use. Mutually exclusive with 'reason'. See detail hangup codes at here. https://docs.asterisk.org/Configuration/Miscellaneous/Hangup-Cause-Mappings/",
"paramType": "query",
"required": false,
"allowMultiple": false,