[svn-commits] trunk r36251 - /trunk/configs/sip.conf.sample
svn-commits at lists.digium.com
svn-commits at lists.digium.com
Thu Jun 29 00:04:44 MST 2006
Author: oej
Date: Thu Jun 29 02:04:43 2006
New Revision: 36251
URL: http://svn.digium.com/view/asterisk?rev=36251&view=rev
Log:
reformatting sip.conf.sample a bit, adding dumphistory that was not documented
Modified:
trunk/configs/sip.conf.sample
Modified: trunk/configs/sip.conf.sample
URL: http://svn.digium.com/view/asterisk/trunk/configs/sip.conf.sample?rev=36251&r1=36250&r2=36251&view=diff
==============================================================================
--- trunk/configs/sip.conf.sample (original)
+++ trunk/configs/sip.conf.sample Thu Jun 29 02:04:43 2006
@@ -25,9 +25,7 @@
[general]
context=default ; Default context for incoming calls
-;allowguest=no ; Allow or reject guest calls (default is yes,
- ; this can also be set to 'osp'
- ; if asterisk was compiled with OSP support.)
+;allowguest=no ; Allow or reject guest calls (default is yes)
allowoverlap=no ; Disable overlap dialing support. (Default is yes)
;allowtransfer=no ; Disable all transfers (unless enabled in peers or users)
; Default is enabled
@@ -49,15 +47,6 @@
; If configured, Asterisk will only allow
; INVITE and REFER to non-local domains
; Use "sip show domains" to list local domains
-;domain=mydomain.tld,mydomain-incoming
- ; Add domain and configure incoming context
- ; for external calls to this domain
-;domain=1.2.3.4 ; Add IP address as local domain
- ; You can have several "domain" settings
-;allowexternalinvites=no ; Disable INVITE and REFER to non-local domains
- ; Default is yes
-;autodomain=yes ; Turn this on to have Asterisk add local host
- ; name and local IP to domain list.
;pedantic=yes ; Enable checking of tags in headers,
; international character conversions in URIs
; and multiline formatted headers for strict
@@ -79,9 +68,6 @@
;vmexten=voicemail ; dialplan extension to reach mailbox sets the
; Message-Account in the MWI notify message
; defaults to "asterisk"
-;recordhistory=yes ; Record SIP history by default
- ; (see sip history / sip no history)
-
;disallow=all ; First disallow all codecs
;allow=ulaw ; Allow codecs in order of preference
;allow=ilbc ;
@@ -114,8 +100,6 @@
; auto : Use rfc2833 if offered, inband otherwise
;compactheaders = yes ; send compact sip headers.
-;sipdebug = yes ; Turn on SIP debugging by default, from
- ; the moment the channel loads this configuration
;
;videosupport=yes ; Turn on support for SIP video
;maxcallbitrate=384 ; Maximum bitrate for video calls (default 384 kb/s)
@@ -123,8 +107,33 @@
; for peers and users as well
;callevents=no ; generate manager events when sip ua
; performs events (e.g. hold)
-
-;--------------------------- STATUS NOTIFICATIONS (SUBSCRIPTIONS) ---------
+;alwaysauthreject = yes ; When an incoming INVITE or REGISTER is to be rejected,
+ ; for any reason, always reject with '401 Unauthorized'
+ ; instead of letting the requester know whether there was
+ ; a matching user or peer for their request
+;
+; If regcontext is specified, Asterisk will dynamically create and destroy a
+; NoOp priority 1 extension for a given peer who registers or unregisters with
+; us and have a "regexten=" configuration item.
+; Multiple contexts may be specified by separating them with '&'. The
+; actual extension is the 'regexten' parameter of the registering peer or its
+; name if 'regexten' is not provided. If more than one context is provided,
+; the context must be specified within regexten by appending the desired
+; context after '@'. More than one regexten may be supplied if they are
+; separated by '&'. Patterns may be used in regexten.
+;
+;regcontext=sipregistrations
+;
+;--------------------------- SIP DEBUGGING ---------------------------------------------------
+;sipdebug = yes ; Turn on SIP debugging by default, from
+ ; the moment the channel loads this configuration
+;recordhistory=yes ; Record SIP history by default
+ ; (see sip history / sip no history)
+;dumphistory=yes ; Dump SIP history at end of SIP dialogue
+ ; SIP history is output to the DEBUG logging channel
+
+
+;--------------------------- STATUS NOTIFICATIONS (SUBSCRIPTIONS) ----------------------------
; You can subscribe to the status of extensions with a "hint" priority
; (See extensions.conf.sample for examples)
; chan_sip support two major formats for notifications: dialog-info and SIMPLE
@@ -136,21 +145,6 @@
; Useful to limit subscriptions to local extensions
; Settable per peer/user also
;notifyringing = yes ; Notify subscriptions on RINGING state
-;alwaysauthreject = yes ; When an incoming INVITE or REGISTER is to be rejected,
- ; for any reason, always reject with '401 Unauthorized'
- ; instead of letting the requester know whether there was
- ; a matching user or peer for their request
-;
-; If regcontext is specified, Asterisk will dynamically create and destroy a
-; NoOp priority 1 extension for a given peer who registers or unregisters with
-; us. Multiple contexts may be specified by separating them with '&'. The
-; actual extension is the 'regexten' parameter of the registering peer or its
-; name if 'regexten' is not provided. If more than one context is provided,
-; the context must be specified within regexten by appending the desired
-; context after '@'. More than one regexten may be supplied if they are
-; separated by '&'. Patterns may be used in regexten.
-;
-;regcontext=sipregistrations
;----------------------------------------- T.38 FAX PASSTHROUGH SUPPORT -----------------------
;
; These settings are available in the [general] section as well as in device configurations
@@ -301,13 +295,23 @@
; To disallow requests for domains not serviced by this server:
; allowexternaldomains=no
-; fromdomain=mydomain.tld ; When making outbound SIP INVITEs to
- ; non-peers, use your primary domain "identity"
- ; for From: headers instead of just your IP
- ; address. This is to be polite and
- ; it may be a mandatory requirement for some
- ; destinations which do not have a prior
- ; account relationship with your server.
+;domain=mydomain.tld,mydomain-incoming
+ ; Add domain and configure incoming context
+ ; for external calls to this domain
+;domain=1.2.3.4 ; Add IP address as local domain
+ ; You can have several "domain" settings
+;allowexternalinvites=no ; Disable INVITE and REFER to non-local domains
+ ; Default is yes
+;autodomain=yes ; Turn this on to have Asterisk add local host
+ ; name and local IP to domain list.
+
+; fromdomain=mydomain.tld ; When making outbound SIP INVITEs to
+ ; non-peers, use your primary domain "identity"
+ ; for From: headers instead of just your IP
+ ; address. This is to be polite and
+ ; it may be a mandatory requirement for some
+ ; destinations which do not have a prior
+ ; account relationship with your server.
;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
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