[asterisk-commits] lmadsen: branch lmadsen/sample_config_refactor r328426 - in /team/lmadsen/sam...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Fri Jul 15 14:21:08 CDT 2011
Author: lmadsen
Date: Fri Jul 15 14:21:03 2011
New Revision: 328426
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=328426
Log:
Add more files to core-configs. Moving along now.
Added:
team/lmadsen/sample_config_refactor/core-configs/calendar.conf (with props)
team/lmadsen/sample_config_refactor/core-configs/ccss.conf (with props)
team/lmadsen/sample_config_refactor/core-configs/chan_dahdi.conf (with props)
team/lmadsen/sample_config_refactor/core-configs/default/calendar.sections.conf (with props)
team/lmadsen/sample_config_refactor/core-configs/default/ccss.general.conf (with props)
team/lmadsen/sample_config_refactor/core-configs/default/chan_dahdi.channels.conf (with props)
team/lmadsen/sample_config_refactor/core-configs/default/chan_dahdi.sections.conf (with props)
team/lmadsen/sample_config_refactor/core-configs/default/chan_dahdi.trunkgroups.conf (with props)
team/lmadsen/sample_config_refactor/core-configs/default/musiconhold.general.conf (with props)
team/lmadsen/sample_config_refactor/core-configs/default/musiconhold.sections.conf (with props)
team/lmadsen/sample_config_refactor/core-configs/default/res_curl.globals.conf (with props)
team/lmadsen/sample_config_refactor/core-configs/default/res_fax.general.conf (with props)
team/lmadsen/sample_config_refactor/core-configs/default/res_odbc.ENV.conf (with props)
team/lmadsen/sample_config_refactor/core-configs/default/res_odbc.sections.conf (with props)
team/lmadsen/sample_config_refactor/core-configs/default/res_stun_monitor.general.conf (with props)
team/lmadsen/sample_config_refactor/core-configs/default/sip.general.conf (with props)
team/lmadsen/sample_config_refactor/core-configs/musiconhold.conf (with props)
team/lmadsen/sample_config_refactor/core-configs/res_curl.conf (with props)
team/lmadsen/sample_config_refactor/core-configs/res_fax.conf (with props)
team/lmadsen/sample_config_refactor/core-configs/res_odbc.conf (with props)
team/lmadsen/sample_config_refactor/core-configs/res_stun_monitor.conf (with props)
team/lmadsen/sample_config_refactor/core-configs/sip.conf (with props)
team/lmadsen/sample_config_refactor/core-configs/site/calendar.sections.conf (with props)
team/lmadsen/sample_config_refactor/core-configs/site/ccss.general.conf (with props)
team/lmadsen/sample_config_refactor/core-configs/site/chan_dahdi.channels.conf (with props)
team/lmadsen/sample_config_refactor/core-configs/site/chan_dahdi.sections.conf (with props)
team/lmadsen/sample_config_refactor/core-configs/site/chan_dahdi.trunkgroups.conf (with props)
team/lmadsen/sample_config_refactor/core-configs/site/musiconhold.general.conf (with props)
team/lmadsen/sample_config_refactor/core-configs/site/musiconhold.sections.conf (with props)
team/lmadsen/sample_config_refactor/core-configs/site/res_curl.globals.conf (with props)
team/lmadsen/sample_config_refactor/core-configs/site/res_fax.general.conf (with props)
team/lmadsen/sample_config_refactor/core-configs/site/res_odbc.ENV.conf (with props)
team/lmadsen/sample_config_refactor/core-configs/site/res_odbc.sections.conf (with props)
team/lmadsen/sample_config_refactor/core-configs/site/res_stun_monitor.general.conf (with props)
team/lmadsen/sample_config_refactor/core-configs/site/sip.general.conf (with props)
Modified:
team/lmadsen/sample_config_refactor/core-configs/site/modules.modules.conf
Added: team/lmadsen/sample_config_refactor/core-configs/calendar.conf
URL: http://svnview.digium.com/svn/asterisk/team/lmadsen/sample_config_refactor/core-configs/calendar.conf?view=auto&rev=328426
==============================================================================
--- team/lmadsen/sample_config_refactor/core-configs/calendar.conf (added)
+++ team/lmadsen/sample_config_refactor/core-configs/calendar.conf Fri Jul 15 14:21:03 2011
@@ -1,0 +1,42 @@
+;[calendar1]
+;type = ical ; type of calendar--currently supported: ical, caldav, exchange, or ews
+;url = https://example.com/home/jdoe/Calendar/ ; URL to shared calendar (Zimbra example)
+;user = jdoe ; web username
+;secret = supersecret ; web password
+;refresh = 15 ; refresh calendar every n minutes
+;timeframe = 60 ; number of minutes of calendar data to pull for each refresh period
+; ; should always be >= refresh
+;
+; You can set up res_calendar to execute a call upon an upcoming busy status
+; The following fields are available from the ${CALENDAR_EVENT(<field>)} dialplan function:
+;
+; summary : The VEVENT Summary property or Exchange subject
+; description : The text description of the vent
+; organizer : The organizer of the event
+; location : The location field of the event
+; calendar : The name of the calendar tied to the event
+; uid : The unique ID for this event
+; start : Start time of the event
+; end : The end time of the event
+; busystate : 0=FREE, 1=TENTATIVE, 2=BUSY
+;
+;autoreminder = 10 ; Override event-defined reminder before each busy status (in mins)
+;
+;channel = SIP/60001 ; Channel to dial
+;context = default ; Context to connect to on answer
+;extension = 123 ; Extension to connect to on answer
+;
+; or
+;
+;app = Playback ; Application to execute on answer (instead of context/extension)
+;appdata = tt-weasels ; Data part of application to execute on answer
+;
+;waittime = 30 ; How long to wait for an answer, defaults to 30 seconds
+;
+; Channel variables can be set on the notification channel. The format is
+; setvar=name=value. Variable subsitution is done on the value to allow the use of dialplan
+; functions like CALENDAR_EVENT. The variables are set in order, so one can use the value
+; of earlier variables in the definition of later ones.
+;
+#include default/calendar.sections.conf
+#include site/calendar.sections.conf
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Added: team/lmadsen/sample_config_refactor/core-configs/ccss.conf
URL: http://svnview.digium.com/svn/asterisk/team/lmadsen/sample_config_refactor/core-configs/ccss.conf?view=auto&rev=328426
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--- team/lmadsen/sample_config_refactor/core-configs/ccss.conf (added)
+++ team/lmadsen/sample_config_refactor/core-configs/ccss.conf Fri Jul 15 14:21:03 2011
@@ -1,0 +1,10 @@
+;
+; --- 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)
+;
+
+[general]
+#include default/ccss.general.conf
+#include site/ccss.general.conf
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Added: team/lmadsen/sample_config_refactor/core-configs/chan_dahdi.conf
URL: http://svnview.digium.com/svn/asterisk/team/lmadsen/sample_config_refactor/core-configs/chan_dahdi.conf?view=auto&rev=328426
==============================================================================
--- team/lmadsen/sample_config_refactor/core-configs/chan_dahdi.conf (added)
+++ team/lmadsen/sample_config_refactor/core-configs/chan_dahdi.conf Fri Jul 15 14:21:03 2011
@@ -1,0 +1,19 @@
+;
+; DAHDI Telephony Configuration file
+;
+; You need to restart Asterisk to re-configure the DAHDI channel
+; CLI> module reload chan_dahdi.so
+; will reload the configuration file, but not all configuration options
+; are re-configured during a reload (signalling, as well as PRI and
+; SS7-related settings cannot be changed on a reload).
+
+[trunkgroups]
+#include default/chan_dahdi.trunkgroups.conf
+#include site/chan_dahdi.trunkgroups.conf
+
+[channels]
+#include default/chan_dahdi.channels.conf
+#include site/chan_dahdi.channels.conf
+
+#include default/chan_dahdi.sections.conf
+#include site/chan_dahdi.sections.conf
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Added: team/lmadsen/sample_config_refactor/core-configs/default/calendar.sections.conf
URL: http://svnview.digium.com/svn/asterisk/team/lmadsen/sample_config_refactor/core-configs/default/calendar.sections.conf?view=auto&rev=328426
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Added: team/lmadsen/sample_config_refactor/core-configs/default/ccss.general.conf
URL: http://svnview.digium.com/svn/asterisk/team/lmadsen/sample_config_refactor/core-configs/default/ccss.general.conf?view=auto&rev=328426
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Added: team/lmadsen/sample_config_refactor/core-configs/default/chan_dahdi.channels.conf
URL: http://svnview.digium.com/svn/asterisk/team/lmadsen/sample_config_refactor/core-configs/default/chan_dahdi.channels.conf?view=auto&rev=328426
==============================================================================
--- team/lmadsen/sample_config_refactor/core-configs/default/chan_dahdi.channels.conf (added)
+++ team/lmadsen/sample_config_refactor/core-configs/default/chan_dahdi.channels.conf Fri Jul 15 14:21:03 2011
@@ -1,0 +1,14 @@
+usecallerid=yes
+callwaiting=yes
+usecallingpres=yes
+callwaitingcallerid=yes
+threewaycalling=yes
+transfer=yes
+canpark=yes
+cancallforward=yes
+callreturn=yes
+echocancel=yes
+echocancelwhenbridged=yes
+group=1
+callgroup=1
+pickupgroup=1
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Added: team/lmadsen/sample_config_refactor/core-configs/default/chan_dahdi.sections.conf
URL: http://svnview.digium.com/svn/asterisk/team/lmadsen/sample_config_refactor/core-configs/default/chan_dahdi.sections.conf?view=auto&rev=328426
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Added: team/lmadsen/sample_config_refactor/core-configs/default/chan_dahdi.trunkgroups.conf
URL: http://svnview.digium.com/svn/asterisk/team/lmadsen/sample_config_refactor/core-configs/default/chan_dahdi.trunkgroups.conf?view=auto&rev=328426
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Added: team/lmadsen/sample_config_refactor/core-configs/default/musiconhold.general.conf
URL: http://svnview.digium.com/svn/asterisk/team/lmadsen/sample_config_refactor/core-configs/default/musiconhold.general.conf?view=auto&rev=328426
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URL: http://svnview.digium.com/svn/asterisk/team/lmadsen/sample_config_refactor/core-configs/default/musiconhold.sections.conf?view=auto&rev=328426
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--- team/lmadsen/sample_config_refactor/core-configs/default/musiconhold.sections.conf (added)
+++ team/lmadsen/sample_config_refactor/core-configs/default/musiconhold.sections.conf Fri Jul 15 14:21:03 2011
@@ -1,0 +1,3 @@
+[default]
+mode=files
+directory=moh
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Added: team/lmadsen/sample_config_refactor/core-configs/default/res_curl.globals.conf
URL: http://svnview.digium.com/svn/asterisk/team/lmadsen/sample_config_refactor/core-configs/default/res_curl.globals.conf?view=auto&rev=328426
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--- team/lmadsen/sample_config_refactor/core-configs/default/res_curl.globals.conf (added)
+++ team/lmadsen/sample_config_refactor/core-configs/default/res_curl.globals.conf Fri Jul 15 14:21:03 2011
@@ -1,0 +1,2 @@
+proxytype=http
+proxyport=8001
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Added: team/lmadsen/sample_config_refactor/core-configs/default/res_fax.general.conf
URL: http://svnview.digium.com/svn/asterisk/team/lmadsen/sample_config_refactor/core-configs/default/res_fax.general.conf?view=auto&rev=328426
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--- team/lmadsen/sample_config_refactor/core-configs/default/res_fax.general.conf (added)
+++ team/lmadsen/sample_config_refactor/core-configs/default/res_fax.general.conf Fri Jul 15 14:21:03 2011
@@ -1,0 +1,5 @@
+maxrate=14400
+minrate=2400
+statusevents=yes
+modems=v17,v27,v29
+ecm=yes
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Added: team/lmadsen/sample_config_refactor/core-configs/default/res_odbc.ENV.conf
URL: http://svnview.digium.com/svn/asterisk/team/lmadsen/sample_config_refactor/core-configs/default/res_odbc.ENV.conf?view=auto&rev=328426
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Added: team/lmadsen/sample_config_refactor/core-configs/default/res_stun_monitor.general.conf
URL: http://svnview.digium.com/svn/asterisk/team/lmadsen/sample_config_refactor/core-configs/default/res_stun_monitor.general.conf?view=auto&rev=328426
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--- team/lmadsen/sample_config_refactor/core-configs/default/res_stun_monitor.general.conf (added)
+++ team/lmadsen/sample_config_refactor/core-configs/default/res_stun_monitor.general.conf Fri Jul 15 14:21:03 2011
@@ -1,0 +1,1 @@
+stunrefresh=30
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Added: team/lmadsen/sample_config_refactor/core-configs/default/sip.general.conf
URL: http://svnview.digium.com/svn/asterisk/team/lmadsen/sample_config_refactor/core-configs/default/sip.general.conf?view=auto&rev=328426
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--- team/lmadsen/sample_config_refactor/core-configs/default/sip.general.conf (added)
+++ team/lmadsen/sample_config_refactor/core-configs/default/sip.general.conf Fri Jul 15 14:21:03 2011
@@ -1,0 +1,9 @@
+context=default ; Default context for incoming calls
+allowguest=yes
+allowoverlap=no ; Disable overlap dialing support. (Default is yes)
+udpbindaddr=0.0.0.0 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all)
+ ; Optionally add a port number, 192.168.1.1:5062 (default is port 5060)
+tcpenable=no ; Enable server for incoming TCP connections (default is no)
+tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces)
+ ; Optionally add a port number, 192.168.1.1:5062 (default is port 5060)
+srvlookup=yes ; Enable DNS SRV lookups on outbound calls
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Added: team/lmadsen/sample_config_refactor/core-configs/musiconhold.conf
URL: http://svnview.digium.com/svn/asterisk/team/lmadsen/sample_config_refactor/core-configs/musiconhold.conf?view=auto&rev=328426
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--- team/lmadsen/sample_config_refactor/core-configs/musiconhold.conf (added)
+++ team/lmadsen/sample_config_refactor/core-configs/musiconhold.conf Fri Jul 15 14:21:03 2011
@@ -1,0 +1,9 @@
+;
+; Music on Hold -- Sample Configuration
+;
+[general]
+#include default/musiconhold.general.conf
+#include site/musiconhold.general.conf
+
+#include default/musiconhold.sections.conf
+#include site/musiconhold.sections.conf
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Added: team/lmadsen/sample_config_refactor/core-configs/res_curl.conf
URL: http://svnview.digium.com/svn/asterisk/team/lmadsen/sample_config_refactor/core-configs/res_curl.conf?view=auto&rev=328426
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--- team/lmadsen/sample_config_refactor/core-configs/res_curl.conf (added)
+++ team/lmadsen/sample_config_refactor/core-configs/res_curl.conf Fri Jul 15 14:21:03 2011
@@ -1,0 +1,6 @@
+;
+; CURLOPT global settings (mostly set for realtime)
+;
+[globals]
+#include default/res_curl.globals.conf
+#include site/res_curl.globals.conf
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Added: team/lmadsen/sample_config_refactor/core-configs/res_fax.conf
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--- team/lmadsen/sample_config_refactor/core-configs/res_fax.conf (added)
+++ team/lmadsen/sample_config_refactor/core-configs/res_fax.conf Fri Jul 15 14:21:03 2011
@@ -1,0 +1,5 @@
+; Generic Fax Application configuration
+
+[general]
+#include default/res_fax.general.conf
+#include site/res_fax.general.conf
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URL: http://svnview.digium.com/svn/asterisk/team/lmadsen/sample_config_refactor/core-configs/res_odbc.conf?view=auto&rev=328426
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--- team/lmadsen/sample_config_refactor/core-configs/res_odbc.conf (added)
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+;;; odbc setup file
+
+; ENV is a global set of environmental variables that will get set.
+; Note that all environmental variables can be seen by all connections,
+; so you can't have different values for different connections.
+[ENV]
+#include default/res_odbc.ENV.conf
+#include site/res_odbc.ENV.conf
+
+#include default/res_odbc.sections.conf
+#include site/res_odbc.sections.conf
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--- team/lmadsen/sample_config_refactor/core-configs/res_stun_monitor.conf (added)
+++ team/lmadsen/sample_config_refactor/core-configs/res_stun_monitor.conf Fri Jul 15 14:21:03 2011
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+;
+; Configuration file for the res_stun_monitor module
+;
+; The res_stun_monitor module sends STUN requests to a configured STUN server
+; periodically. If the monitor detects a change in the external ip or port
+; provided by the STUN server an event is sent out internally within Asterisk
+; to alert all listeners to that event of the change.
+
+; The current default listeners for the netork change event include chan_sip
+; and chan_iax. Both of these channel drivers by default react to this event
+; by renewing all outbound registrations. This allows the endpoints Asterisk
+; is registering with to become aware of the address change and know the new
+; location.
+;
+[general]
+#include default/res_stun_monitor.general.conf
+#include site/res_stun_monitor.general.conf
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--- team/lmadsen/sample_config_refactor/core-configs/sip.conf (added)
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+;
+; SIP Configuration example for Asterisk
+;
+; Note: Please read the security documentation for Asterisk in order to
+; understand the risks of installing Asterisk with the sample
+; configuration. If your Asterisk is installed on a public
+; IP address connected to the Internet, you will want to learn
+; about the various security settings BEFORE you start
+; Asterisk.
+;
+; Especially note the following settings:
+; - allowguest (default enabled)
+; - permit/deny - IP address filters
+; - contactpermit/contactdeny - IP address filters for registrations
+; - context - Which set of services you offer various users
+[general]
+#include default/sip.general.conf
+#include site/sip.general.conf
+
+[authentication]
+; Global credentials for outbound calls, i.e. when a proxy challenges your
+; Asterisk server for authentication. These credentials override
+; any credentials in peer/register definition if realm is matched.
+;
+; This way, Asterisk can authenticate for outbound calls to other
+; realms. We match realm on the proxy challenge and pick an set of
+; credentials from this list
+; Syntax:
+; auth = <user>:<secret>@<realm>
+; auth = <user>#<md5secret>@<realm>
+; Example:
+;auth=mark:topsecret at digium.com
+;
+; You may also add auth= statements to [peer] definitions
+; Peer auth= override all other authentication settings if we match on realm
+
+;------------------------------------------------------------------------------
+; DEVICE CONFIGURATION
+;
+; The SIP channel has two types of devices, the friend and the peer.
+; * The type=friend is a device type that accepts both incoming and outbound calls,
+; where Asterisk match on the From: username on incoming calls.
+; (A synonym for friend is "user"). This is a type you use for your local
+; SIP phones.
+; * The type=peer also handles both incoming and outbound calls. On inbound calls,
+; Asterisk only matches on IP/port, not on names. This is mostly used for SIP
+; trunks.
+;
+; Use remotesecret for outbound authentication, and secret for authenticating
+; inbound requests. For historical reasons, if no remotesecret is supplied for an
+; outbound registration or call, the secret will be used.
+;
+; For device names, we recommend using only a-z, numerics (0-9) and underscore
+;
+; For local phones, type=friend works most of the time
+;
+; If you have one-way audio, you probably have NAT problems.
+; If Asterisk is on a public IP, and the phone is inside of a NAT device
+; you will need to configure nat option for those phones.
+; Also, turn on qualify=yes to keep the nat session open
+;
+; Configuration options available
+; --------------------
+; context
+; callingpres
+; permit
+; deny
+; secret
+; md5secret
+; remotesecret
+; transport
+; dtmfmode
+; directmedia
+; nat
+; callgroup
+; pickupgroup
+; language
+; allow
+; disallow
+; insecure
+; trustrpid
+; progressinband
+; promiscredir
+; useclientcode
+; accountcode
+; setvar
+; callerid
+; amaflags
+; callcounter
+; busylevel
+; allowoverlap
+; allowsubscribe
+; allowtransfer
+; ignoresdpversion
+; subscribecontext
+; template
+; videosupport
+; maxcallbitrate
+; rfc2833compensate
+; mailbox
+; session-timers
+; session-expires
+; session-minse
+; session-refresher
+; t38pt_usertpsource
+; regexten
+; fromdomain
+; fromuser
+; host
+; port
+; qualify
+; defaultip
+; defaultuser
+; rtptimeout
+; rtpholdtimeout
+; sendrpid
+; outboundproxy
+; rfc2833compensate
+; callbackextension
+; registertrying
+; timert1
+; timerb
+; qualifyfreq
+; t38pt_usertpsource
+; contactpermit ; Limit what a host may register as (a neat trick
+; contactdeny ; is to register at the same IP as a SIP provider,
+; ; then call oneself, and get redirected to that
+; ; same location).
+; directmediapermit
+; directmediadeny
+; unsolicited_mailbox
+; use_q850_reason
+; maxforwards
+; encryption
+; description ; Used to provide a description of the peer in console output
+
+;[sip_proxy]
+; For incoming calls only. Example: FWD (Free World Dialup)
+; We match on IP address of the proxy for incoming calls
+; since we can not match on username (caller id)
+;type=peer
+;context=from-fwd
+;host=fwd.pulver.com
+
+;[sip_proxy-out]
+;type=peer ; we only want to call out, not be called
+;remotesecret=guessit ; Our password to their service
+;defaultuser=yourusername ; Authentication user for outbound proxies
+;fromuser=yourusername ; Many SIP providers require this!
+;fromdomain=provider.sip.domain
+;host=box.provider.com
+;transport=udp,tcp ; This sets the default transport type to udp for outgoing, and will
+; ; accept both tcp and udp. The default transport type is only used for
+; ; outbound messages until a Registration takes place. During the
+; ; peer Registration the transport type may change to another supported
+; ; type if the peer requests so.
+
+;usereqphone=yes ; This provider requires ";user=phone" on URI
+;callcounter=yes ; Enable call counter
+;busylevel=2 ; Signal busy at 2 or more calls
+;outboundproxy=proxy.provider.domain ; send outbound signaling to this proxy, not directly to the peer
+;port=80 ; The port number we want to connect to on the remote side
+ ; Also used as "defaultport" in combination with "defaultip" settings
+
+;--- sample definition for a provider
+;[provider1]
+;type=peer
+;host=sip.provider1.com
+;fromuser=4015552299 ; how your provider knows you
+;remotesecret=youwillneverguessit ; The password we use to authenticate to them
+;secret=gissadetdu ; The password they use to contact us
+;callbackextension=123 ; Register with this server and require calls coming back to this extension
+;transport=udp,tcp ; This sets the transport type to udp for outgoing, and will
+; ; accept both tcp and udp. Default is udp. The first transport
+; ; listed will always be used for outgoing connections.
+;unsolicited_mailbox=4015552299 ; If the remote SIP server sends an unsolicited MWI NOTIFY message the new/old
+; ; message count will be stored in the configured virtual mailbox. It can be used
+; ; by any device supporting MWI by specifying <configured value>@SIP_Remote as the
+; ; mailbox.
+
+;
+; Because you might have a large number of similar sections, it is generally
+; convenient to use templates for the common parameters, and add them
+; the the various sections. Examples are below, and we can even leave
+; the templates uncommented as they will not harm:
+
+[basic-options](!) ; a template
+ dtmfmode=rfc2833
+ context=from-office
+ type=friend
+
+[natted-phone](!,basic-options) ; another template inheriting basic-options
+ nat=yes
+ directmedia=no
+ host=dynamic
+
+[public-phone](!,basic-options) ; another template inheriting basic-options
+ nat=no
+ directmedia=yes
+
+[my-codecs](!) ; a template for my preferred codecs
+ disallow=all
+ allow=ilbc
+ allow=g729
+ allow=gsm
+ allow=g723
+ allow=ulaw
+
+[ulaw-phone](!) ; and another one for ulaw-only
+ disallow=all
+ allow=ulaw
+
+; and finally instantiate a few phones
+;
+; [2133](natted-phone,my-codecs)
+; secret = peekaboo
+; [2134](natted-phone,ulaw-phone)
+; secret = not_very_secret
+; [2136](public-phone,ulaw-phone)
+; secret = not_very_secret_either
+; ...
+;
+
+; Standard configurations not using templates look like this:
+;
+;[grandstream1]
+;type=friend
+;context=from-sip ; Where to start in the dialplan when this phone calls
+;callerid=John Doe <1234> ; Full caller ID, to override the phones config
+ ; on incoming calls to Asterisk
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