mirror of
https://github.com/asterisk/asterisk.git
synced 2025-10-12 15:45:18 +00:00
Improve sample configuration files (bug #1125)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@3057 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Mark Spencer
@@ -96,6 +96,12 @@ TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0)
|
||||
[iaxtel700]
|
||||
exten => _91700XXXXXXX,1,Dial(IAX2/${IAXINFO}@iaxtel.com/${EXTEN:1}@iaxtel)
|
||||
|
||||
;
|
||||
; The SWITCH statement permits a server to share the dialplain with
|
||||
; another server. Use with care: Reciprocal switch statements are not
|
||||
; allowed (e.g. both A -> B and B -> A), and the switched server needs
|
||||
; to be on-line or else dialing can be severly delayed.
|
||||
;
|
||||
[iaxprovider]
|
||||
;switch => IAX2/user:[key]@myserver/mycontext
|
||||
|
||||
@@ -276,17 +282,29 @@ exten => 8500,2,Goto(s,6)
|
||||
;
|
||||
include => demo
|
||||
|
||||
;
|
||||
; Extensions like the two below can be used for FWD, Nikotel, sipgate etc.
|
||||
; Note that you must have a [sipprovider] section in sip.conf whereas
|
||||
; the otherprovider.net example does not require such a peer definition
|
||||
;
|
||||
;exten => _41X.,1,Dial(SIP/${EXTEN:2}@sipprovider,,r)
|
||||
;exten => _42X.,1,Dial(SIP/user:passwd@${EXTEN:2}@otherprovider.net,30,rT)
|
||||
|
||||
; Real extensions would go here. Generally you want real extensions to be 4 or 5
|
||||
; Real extensions would go here. Generally you want real extensions to be 4 or 5
|
||||
; digits long (although there is no such requirement) and start with a single
|
||||
; digit that is fairly large (like 6 or 7) so that you have plenty of room to
|
||||
; overlap extensions and menu options without conflict. You can alias them with
|
||||
; names, too and use global variables
|
||||
|
||||
;exten => 6245,1,Dial(SIP/Grandstream1,20,rt) ; permit transfer
|
||||
;exten => 6245,1,Dial(SIP/Grandstream1&SIP/Xlite1,20,rtT)
|
||||
;exten => 6361,1,Dial(IAX2/JaneDoe,,rm) ; ring without time limit
|
||||
;exten => 6389,1,Dial(MGCP/aaln/1@192.168.0.14)
|
||||
;exten => 6394,1,Dial(Local/6275/n) ; this will dial ${MARK}
|
||||
|
||||
;exten => 6275,1,Macro(stdexten,6275,${MARK}) ; assuming ${MARK} is something like Zap/2
|
||||
;exten => mark,1,Goto(6275|1) ; alias mark to 6275
|
||||
;exten => 6236,1,Macro(stdexten,6236,${WIL}) ; Ditto for wil
|
||||
;exten => 6275,1,Macro(stdexten,6275,${MARK}) ; assuming ${MARK} is something like Zap/2
|
||||
;exten => mark,1,Goto(6275|1) ; alias mark to 6275
|
||||
;exten => 6536,1,Macro(stdexten,6236,${WIL}) ; Ditto for wil
|
||||
;exten => wil,1,Goto(6236|1)
|
||||
;
|
||||
; Some other handy things are an extension for checking voicemail via
|
||||
@@ -297,7 +315,7 @@ include => demo
|
||||
;
|
||||
; Or a conference room (you'll need to edit meetme.conf to enable this room)
|
||||
;
|
||||
;exten => 8600,1,Meetme,1234
|
||||
;exten => 8600,1,Meetme(1234)
|
||||
;
|
||||
; Or playing an announcement to the called party, as soon it answers
|
||||
;
|
||||
|
||||
@@ -1,5 +1,5 @@
|
||||
;
|
||||
; Internet Phone Jack
|
||||
; isdn4linux
|
||||
;
|
||||
; Configuration file
|
||||
;
|
||||
@@ -11,7 +11,8 @@ context=remote
|
||||
;
|
||||
; Modem Drivers to load
|
||||
;
|
||||
driver=aopen
|
||||
driver=aopen ; modem by AOpen
|
||||
;driver=i4l ; isdn4linux - an alternative to i4l is to use chan_capi
|
||||
;
|
||||
; Default language
|
||||
;
|
||||
@@ -26,7 +27,7 @@ driver=aopen
|
||||
; We can strip a given number of digits on outgoing dialing, so, for example
|
||||
; you can have it dial "8871042" when given "98871042".
|
||||
;
|
||||
stripmsd=1
|
||||
stripmsd=0
|
||||
;
|
||||
; Type of dialing
|
||||
;
|
||||
@@ -45,7 +46,7 @@ mode=immediate
|
||||
;
|
||||
;device => /dev/ttyS3
|
||||
;
|
||||
; ISDN example
|
||||
; ISDN example (using i4l)
|
||||
;
|
||||
;msn=39907835
|
||||
;device => /dev/ttyI0
|
||||
|
||||
@@ -21,25 +21,34 @@
|
||||
;
|
||||
|
||||
[general]
|
||||
port = 5060 ; Port to bind to
|
||||
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
|
||||
context=default ; Default context for incoming calls
|
||||
;realm=mydomain.tld ; Realm for digest authentication
|
||||
; defaults to "asterisk"
|
||||
; Realms MUST be globally unique according to RFC 3261
|
||||
; Set this to your host name or domain name
|
||||
port=5060 ; UDP Port to bind to (SIP standard port is 5060)
|
||||
bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
|
||||
;srvlookup=yes ; Enable DNS SRV lookups on outbound calls
|
||||
; Note: Asterisk only uses the first host in SRV records
|
||||
;pedantic=yes ; Enable slow, pedantic checking for Pingtel
|
||||
; and multiline formatted headers for strict
|
||||
; SIP compatibility
|
||||
;tos=lowdelay ; IP QoS parameter, either keyword or value
|
||||
; like tos=184
|
||||
;tos=184 ; Set IP QoS to either a keyword or numeric val
|
||||
;tos=lowdelay ; lowdelay,throughput,reliability,mincost,none
|
||||
;maxexpirey=3600 ; Max length of incoming registration we allow
|
||||
;realm=asterisk ; Our global authentication realm
|
||||
;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
|
||||
;disallow=all ; First disallow all codecs
|
||||
;allow=ulaw ; Allow codecs in order of preference
|
||||
;allow=ilbc
|
||||
;allow=ilbc ; Note: codec order is respected only in [general]
|
||||
;musicclass=default ; Sets the default music on hold class for all SIP calls
|
||||
; This may also be set for individual users/peers
|
||||
;language=en ; Default language setting for all users/peers
|
||||
; This may also be set for individual users/peers
|
||||
;relaxdtmf=yes ; Relax dtmf handling
|
||||
|
||||
|
||||
; Asterisk can register as a SIP user agent to a SIP proxy (provider)
|
||||
; Format for the register statement is:
|
||||
@@ -56,14 +65,17 @@ context = default ; Default context for incoming calls
|
||||
;
|
||||
;register => 1234:password@mysipprovider.com
|
||||
;
|
||||
; Will call to the 's' extension
|
||||
; This will pass incoming calls to the 's' extension
|
||||
;
|
||||
;
|
||||
;register => 2345@mysipprovider.com/1234
|
||||
;register => 2345:password@sip_proxy/1234
|
||||
;
|
||||
; Register 2345 at sip provider. Calls from this provider connect to local
|
||||
; Register 2345 at sip provider 'sip_proxy'. Calls from this provider connect to local
|
||||
; extension 1234 in extensions.conf default context, unless you define
|
||||
; [mysipprovider.com] in a section below, and configure a context
|
||||
; unless you configure a [sip_proxy] section below, and configure a context.
|
||||
; Tip 1: Avoid assigning hostname to a sip.conf section like [provider.com]
|
||||
; Tip 2: Use separate type=peer and type=user sections for SIP providers
|
||||
; (instead of type=friend) if you have calls in both directions
|
||||
|
||||
|
||||
;externip = 200.201.202.203 ; Address that we're going to put in outbound SIP messages
|
||||
@@ -76,51 +88,143 @@ context = default ; Default context for incoming calls
|
||||
; 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
|
||||
;localnet=169.254.0.0/255.255.0.0 ;Zero conf local network
|
||||
|
||||
;[snomsip]
|
||||
;-----------------------------------------------------------------------------------
|
||||
; Users and peers have different settings available. Friends have all settings,
|
||||
; since a friend is both a peer and a user
|
||||
;
|
||||
; User config options: Peer configuration:
|
||||
; -------------------- -------------------
|
||||
; context context
|
||||
; permit permit
|
||||
; deny deny
|
||||
; auth auth
|
||||
; secret secret
|
||||
; md5secret md5secret
|
||||
; dtmfmode dtmfmode
|
||||
; canreinvite canreinvite
|
||||
; nat nat
|
||||
; callgroup callgroup
|
||||
; pickupgroup pickupgroup
|
||||
; language language
|
||||
; allow allow
|
||||
; disallow disallow
|
||||
; insecure insecure
|
||||
; callerid
|
||||
; accountcode
|
||||
; amaflags
|
||||
; incominglimit
|
||||
; outgoinglimit
|
||||
; restrictcid
|
||||
; mailbox
|
||||
; username
|
||||
; template
|
||||
; fromdomain
|
||||
; fromuser
|
||||
; host
|
||||
; mask
|
||||
; port
|
||||
; qualify
|
||||
; defaultip
|
||||
|
||||
|
||||
;[sip_proxy]
|
||||
; For incoming calls only. Example: FWD (Free World Dialup)
|
||||
;type=user
|
||||
;context=from-fwd
|
||||
|
||||
;[sip_proxy-out]
|
||||
;type=peer ; we only want to call out, not be called
|
||||
;secret=guessit
|
||||
;username=yourusername
|
||||
;fromuser=yourusername ; Many SIP providers require this!
|
||||
;host=box.provider.com
|
||||
|
||||
;[grandstream1]
|
||||
;type=friend ; either "friend" (peer+user), "peer" or "user"
|
||||
;context=from-sip
|
||||
;username=grandstream1 ; usually matches the [section] title
|
||||
;fromuser=grandstream1 ; overrides the callerid, e.g. required by FWD
|
||||
;callerid=John Doe <1234>
|
||||
;host=192.168.0.23 ; we have a static but private IP address
|
||||
;nat=no ; there is not NAT between phone and Asterisk
|
||||
;canreinvite=yes ; allow RTP voice traffic to bypass Asterisk
|
||||
;dtmfmode=info ; either RFC2833 or INFO for the BudgeTone
|
||||
;outgoinglimit=1 ; disable callwaiting signal (2nd call to phone)
|
||||
;incominglimit=1 ; permit only 1 outgoing call at a time
|
||||
;mailbox=1234@default ; mailbox 1234 in voicemail context "default"
|
||||
;disallow=all ; need to disallow=all before we can use allow=
|
||||
;allow=ulaw ; Note: In user sections the order of codecs
|
||||
; listed with allow= does NOT matter!
|
||||
;allow=alaw
|
||||
;allow=g723.1 ; Asterisk only supports g723.1 pass-thru!
|
||||
;allow=g729 ; Pass-thru only unless g729 license obtained
|
||||
|
||||
|
||||
;[xlite1]
|
||||
;Turn off silence suppression in X-Lite ("Transmit Silence"=YES)!
|
||||
;Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed
|
||||
;type=friend
|
||||
;secret=blah
|
||||
;username=xlite1
|
||||
;callerid="Jane Smith" <5678>
|
||||
;host=dynamic
|
||||
;nat=yes ; X-Lite is behind a NAT router
|
||||
;canreinvite=no ; Typically set to NO if behind NAT
|
||||
;disallow=all
|
||||
;allow=gsm ; GSM consumes far less bandwidth than ulaw
|
||||
;allow=ulaw
|
||||
;allow=alaw
|
||||
|
||||
|
||||
;[snom]
|
||||
;type=friend ; Friends place calls and receive calls
|
||||
;context=from-sip ; Context for incoming calls from this user
|
||||
;secret=blah
|
||||
;host=dynamic ; This peer register with us
|
||||
;dtmfmode=inband ; Choices are inband, rfc2833, or info
|
||||
;defaultip=192.168.0.59
|
||||
;mailbox=1234,2345 ; Mailbox for message waiting indicator
|
||||
;defaultip=192.168.0.59 ; IP used until peer registers
|
||||
;mailbox=1234,2345 ; Mailboxes for message waiting indicator
|
||||
;restrictcid=yes ; To have the callerid restriced -> sent as ANI
|
||||
;insecure=yes ; To match a peer based by IP address only and not peer
|
||||
;insecure=very ; To allow registered hosts to call without re-authenticating
|
||||
;disallow=all
|
||||
;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw!
|
||||
;mailbox=1234@context,2345 ; Mailbox(-es) for message waiting indicator
|
||||
|
||||
|
||||
;[pingtel]
|
||||
;type=friend
|
||||
;username=pingtel
|
||||
;secret=blah
|
||||
;host=dynamic
|
||||
;insecure=yes ; To match a peer based by IP address only and not peer
|
||||
;insecure=very ; To allow registered hosts to call without re-authenticating
|
||||
;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 ; We are in caller groups 1,3,4
|
||||
;pickupgroup=1,3-5 ; We can do call pick-p for call group 1,3,4,5
|
||||
;defaultip=192.168.0.60 ; IP address to use if peer has not registred
|
||||
|
||||
;callgroup=1,3-4
|
||||
;pickupgroup=1,3-4
|
||||
;defaultip=192.168.0.60
|
||||
|
||||
;[cisco]
|
||||
;[cisco1]
|
||||
;type=friend
|
||||
;username=cisco
|
||||
;username=cisco1
|
||||
;secret=blah
|
||||
;qualify=200 ; Qualify peer is no more than 200ms away
|
||||
;nat=yes ; This phone may be natted
|
||||
; Use IP address that packet is received from
|
||||
; instead of trusting SIP headers
|
||||
;host=dynamic
|
||||
; Send SIP and RTP to IP address that packet is
|
||||
; received from instead of trusting SIP headers
|
||||
;host=dynamic ; This device registers with us
|
||||
;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
|
||||
; behind a NAT).
|
||||
;qualify=200 ; Qualify peer is no more than 200ms away
|
||||
;defaultip=192.168.0.4
|
||||
|
||||
;[cisco1]
|
||||
;[cisco2]
|
||||
;type=friend
|
||||
;username=cisco1
|
||||
;username=cisco2
|
||||
;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
|
||||
|
||||
Reference in New Issue
Block a user