From 023e27f695a5f20077bcf5654ecdf32f4b73ed53 Mon Sep 17 00:00:00 2001 From: Olle Johansson Date: Thu, 6 Apr 2006 15:23:14 +0000 Subject: [PATCH] Formatting fixes git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@17861 65c4cc65-6c06-0410-ace0-fbb531ad65f3 --- configs/sip.conf.sample | 53 +++++++++++++++++++++++------------------ 1 file changed, 30 insertions(+), 23 deletions(-) diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample index 87d2a073c4..3133feee0a 100644 --- a/configs/sip.conf.sample +++ b/configs/sip.conf.sample @@ -25,7 +25,8 @@ [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) @@ -64,7 +65,8 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ;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 @@ -120,7 +122,8 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ;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 @@ -162,16 +165,16 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ;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 @@ -183,8 +186,8 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; 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 @@ -225,26 +228,27 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ;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|) - ; 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' @@ -360,12 +364,14 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; 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. @@ -459,7 +465,8 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ;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