Improve sample configuration files (bug #1125)

git-svn-id: http://svn.digium.com/svn/asterisk/trunk@3057 f38db490-d61c-443f-a65b-d21fe96a405b

3 files changed, 165 insertions, 42 deletions

diff --git a/configs/extensions.conf.sample b/configs/extensions.conf.sample
index 33456d3db..a93bdb42e 100755
--- a/configs/extensions.conf.sample
+++ b/configs/extensions.conf.sample
@@ -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
 ;
diff --git a/configs/modem.conf.sample b/configs/modem.conf.sample
index d177502df..81e3a0ff2 100755
--- a/configs/modem.conf.sample
+++ b/configs/modem.conf.sample
@@ -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
diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample
index c5e7146cf..c314b7099 100755
--- a/configs/sip.conf.sample
+++ b/configs/sip.conf.sample
@@ -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