From b79b6aba5d51a55037aab9480b1acae3a2d72ceb Mon Sep 17 00:00:00 2001 From: Malcolm Davenport <malcolmd@digium.com> Date: Fri, 19 Mar 2004 20:30:03 +0000 Subject: [PATCH] Bug # 1013: More explanation in the sip.conf.sample thanks to oej git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@2476 65c4cc65-6c06-0410-ace0-fbb531ad65f3 --- configs/sip.conf.sample | 70 ++++++++++++++++++++++++++++++++--------- 1 file changed, 56 insertions(+), 14 deletions(-) diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample index ec965e9e3d..d8de566ec8 100755 --- a/configs/sip.conf.sample +++ b/configs/sip.conf.sample @@ -1,28 +1,57 @@ ; ; SIP Configuration for Asterisk ; +; Syntax for specifying a SIP device in extensions.conf is +; SIP/devicename where devicename is defined in a section below. +; +; You may also use +; SIP/username@domain to call any SIP user on the Internet +; (Don't forget to enable DNS SRV records if you want to use this) +; +; If you define a SIP proxy as a peer below, you may call +; SIP/proxyhostname/user or SIP/user@proxyhostname +; where the proxyhostname is defined in a section below +; +; Useful CLI commands to check peers/users: +; sip show peers Show all SIP peers (including friends) +; sip show users Show all SIP users (including friends) +; sip show registry Show status of hosts we register with +; +; sip debug Show all SIP messages +; + [general] port = 5060 ; Port to bind to -bindaddr = 0.0.0.0 ; Address to bind to -;externip = 200.201.202.203 ; Address that we're going to put in SIP messages if we're behind a NAT -;localnet = 192.168.1.0 ; Internal NETWORK address -;localmask = 255.255.255.0 ; Internal netmask -context = default ; Default for incoming calls -;srvlookup = yes ; Enable SRV lookups on outbound calls +bindaddr = 0.0.0.0 ; Address to bind SIP channel to +context = default ; Default context for incoming calls +;srvlookup = yes ; Enable DNS SRV lookups on outbound calls + ; Asterisk only uses the first host in SRV records ;pedantic = yes ; Enable slow, pedantic checking for Pingtel -;tos=lowdelay -;tos=184 +;tos=lowdelay ; IP QoS parameter, either keyword or value + ; like tos=184 ;maxexpirey=3600 ; Max length of incoming registration we allow ;defaultexpirey=120 ; Default length of incoming/outoing registration ;notifymimetype=text/plain ; Allow overriding of mime type in NOTIFY ;videosupport=yes ; Turn on support for SIP video + ;disallow=all ; Disallow all codecs ;allow=ulaw ; Allow codecs in order of preference ;allow=ilbc -; -;register => 1234@mysipprovider.com ; Register with a SIP provider -;register => 2345@mysipprovider.com/1234 ; Register 2345 at sip provider as 1234 here. -; + +;register => 1234:password@mysipprovider.com +;Register with a SIP provider + +;register => 2345@mysipprovider.com/1234 +;Register 2345 at sip provider. Calls from this provider connect to local extension 1234 in extensions.conf. + +;externip = 200.201.202.203 ; Address that we're going to put in outbound SIP messages + ; if we're behind a NAT +;localnet = 192.168.1.0 ; Internal NETWORK address +;localmask = 255.255.255.0 ; Internal netmask + ; The externip, localnet and localmask is used + ; when registering and communication with other proxies + ; that we're registred with + ;[snomsip] ;type=friend ;secret=blah @@ -38,6 +67,9 @@ context = default ; Default for incoming calls ;secret=blah ;host=dynamic ;qualify=1000 ; Consider it down if it's 1 second to reply + ; Helps with NAT session + ; qualify=yes uses default value + ;callgroup=1,3-4 ;pickupgroup=1,3-4 ;defaultip=192.168.0.60 @@ -47,8 +79,14 @@ context = default ; Default for incoming calls ;username=cisco ;secret=blah ;nat=yes ; This phone may be natted + ; Use IP address that packet is received from + ; instead of trusting SIP headers ;host=dynamic -;canreinvite=no ; Cisco poops on reinvite sometimes +;canreinvite=no ; Asterisk by default tries to redirect the + ; RTP media stream (audio) to go directly from + ; the caller to the callee. Some devices do not + ; support this (especially if one of them is + ; behinda a NAT). ;qualify=200 ; Qualify peer is no more than 200ms away ;defaultip=192.168.0.4 @@ -56,8 +94,12 @@ context = default ; Default for incoming calls ;type=friend ;username=cisco1 ;fromuser=markster ; Specify user to put in "from" instead of callerid +;fromdomain=yourdomain.com ; Specify domain to put in "from" instead of callerid + ; fromuser and fromdomain are used when Asterisk + ; places calls to this account. It is not used for + ; calls from this account. ;secret=blah ;host=dynamic ;defaultip=192.168.0.4 ;amaflags=default ; Choices are default, omit, billing, documentation -;accountcode=markster ; Users may be associated with an accountcode tp ease billing +;accountcode=markster ; Users may be associated with an accountcode to ease billing -- GitLab