[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'
+;allowguest=no ; Allow or reject guest calls (default is yes,
+ ; this can also be set to 'osp'
; if asterisk was compiled with OSP support.)
allowoverlap=no ; Disable overlap dialing support. (Default is yes)
;allowsubscribe=no ; Disable support for subscriptions. (Default is yes)
;tos_audio=ef ; Sets TOS for RTP audio packets.
;tos_video=af41 ; Sets TOS for RTP video packets.
-;maxexpiry=3600 ; Max length of incoming registrations/subscriptions we allow (seconds)
+;maxexpiry=3600 ; Maximum allowed time of incoming registrations
+ ; and subscriptions (seconds)
;minexpiry=60 ; Minimum length of registrations/subscriptions (default 60)
;defaultexpiry=120 ; Default length of incoming/outoing registration
;t1min=100 ; Minimum roundtrip time for messages to monitored hosts
;maxcallbitrate=384 ; Maximum bitrate for video calls (default 384 kb/s)
; Videosupport and maxcallbitrate is settable
; for peers and users as well
-;callevents=no ; generate manager events when sip ua performs events (e.g. hold)
+;callevents=no ; generate manager events when sip ua
+ ; performs events (e.g. hold)
;
; If regcontext is specified, Asterisk will dynamically create and destroy a
;registertimeout=20 ; retry registration calls every 20 seconds (default)
;registerattempts=10 ; Number of registration attempts before we give up
- ; 0 = continue forever, hammering the other server until it
- ; accepts the registration
+ ; 0 = continue forever, hammering the other server
+ ; until it accepts the registration
; Default is 0 tries, continue forever
;----------------------------------------- NAT SUPPORT ------------------------
; The externip, externhost and localnet settings are used if you use Asterisk
; behind a NAT device to communicate with services on the outside.
-;externip = 200.201.202.203 ; Address that we're going to put in outbound SIP messages
- ; if we're behind a NAT
+;externip = 200.201.202.203 ; Address that we're going to put in outbound SIP
+ ; messages if we're behind a NAT
; The externip and localnet is used
; when registering and communicating with other proxies
; environments! Use externip instead
;externrefresh=10 ; How often to refresh externhost if
; used
- ; You may add multiple local networks. A reasonable set of defaults
- ; are:
+ ; You may add multiple local networks. A reasonable
+ ; set of defaults are:
;localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local networks
;localnet=10.0.0.0/255.0.0.0 ; Also RFC1918
;localnet=172.16.0.0/12 ; Another RFC1918 with CIDR notation
;rtupdate=yes ; Send registry updates to database using realtime? (yes|no)
; If set to yes, when a SIP UA registers successfully, the ip address,
; the origination port, the registration period, and the username of
- ; the UA will be set to database via realtime. If not present, defaults to 'yes'.
+ ; the UA will be set to database via realtime.
+ ; If not present, defaults to 'yes'.
;rtautoclear=yes ; Auto-Expire friends created on the fly on the same schedule
; as if it had just registered? (yes|no|<seconds>)
- ; If set to yes, when the registration expires, the friend will vanish from
- ; the configuration until requested again. If set to an integer,
- ; friends expire within this number of seconds instead of the
- ; registration interval.
+ ; If set to yes, when the registration expires, the friend will
+ ; vanish from the configuration until requested again. If set
+ ; to an integer, friends expire within this number of seconds
+ ; instead of the registration interval.
;ignoreregexpire=yes ; Enabling this setting has two functions:
;
- ; For non-realtime peers, when their registration expires, the information
- ; will _not_ be removed from memory or the Asterisk database; if you attempt
- ; to place a call to the peer, the existing information will be used in spite
- ; of it having expired
+ ; For non-realtime peers, when their registration expires, the
+ ; information will _not_ be removed from memory or the Asterisk database
+ ; if you attempt to place a call to the peer, the existing information
+ ; will be used in spiteof it having expired
;
; For realtime peers, when the peer is retrieved from realtime storage,
; the registration information will be used regardless of whether
- ; it has expired or not; if it expires while the realtime peer is still in
- ; memory (due to caching or other reasons), the information will not be
- ; removed from realtime storage
+ ; it has expired or not; if it expires while the realtime peer
+ ; is still in memory (due to caching or other reasons), the
+ ; information will not be removed from realtime storage
;----------------------------------------- SIP DOMAIN SUPPORT ------------------------
; Incoming INVITE and REFER messages can be matched against a list of 'allowed'
; since they are not stored in-memory
;------------------------------------------------------------------------------
-; Definitions of locally connected SIP phones
+; Definitions of locally connected SIP devices
;
; type = user a device that authenticates to us by "from" field to place calls
; type = peer a device we place calls to or that calls us and we match by host
; type = friend two configurations (peer+user) in one
;
+; 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 propably have NAT problems.
;type=friend
;secret=blah
;host=dynamic
-;insecure=port ; Allow matching of peer by IP address without matching port number
+;insecure=port ; Allow matching of peer by IP address without
+ ; matching port number
;insecure=invite ; Do not require authentication of incoming INVITEs
;insecure=port,invite ; (both)
;qualify=1000 ; Consider it down if it's 1 second to reply
projects / thirdparty / asterisk.git / commitdiff
