[Asterisk-cvs] asterisk/configs extensions.conf.sample,1.21,1.22 modem.conf.sample,1.6,1.7 sip.conf.sample,1.27,1.28
markster at lists.digium.com
markster at lists.digium.com
Mon May 24 10:55:43 CDT 2004
Update of /usr/cvsroot/asterisk/configs
In directory mongoose.digium.com:/tmp/cvs-serv12668/configs
Modified Files:
extensions.conf.sample modem.conf.sample sip.conf.sample
Log Message:
Improve sample configuration files (bug #1125)
Index: extensions.conf.sample
===================================================================
RCS file: /usr/cvsroot/asterisk/configs/extensions.conf.sample,v
retrieving revision 1.21
retrieving revision 1.22
diff -u -d -r1.21 -r1.22
--- extensions.conf.sample 26 Apr 2004 05:47:45 -0000 1.21
+++ extensions.conf.sample 24 May 2004 15:09:34 -0000 1.22
@@ -96,6 +96,12 @@
[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 @@
;
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 at 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 @@
;
; 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
;
Index: modem.conf.sample
===================================================================
RCS file: /usr/cvsroot/asterisk/configs/modem.conf.sample,v
retrieving revision 1.6
retrieving revision 1.7
diff -u -d -r1.6 -r1.7
--- modem.conf.sample 22 Oct 2002 15:31:47 -0000 1.6
+++ modem.conf.sample 24 May 2004 15:09:34 -0000 1.7
@@ -1,5 +1,5 @@
;
-; Internet Phone Jack
+; isdn4linux
;
; Configuration file
;
@@ -11,7 +11,8 @@
;
; 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 @@
; 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 @@
;
;device => /dev/ttyS3
;
-; ISDN example
+; ISDN example (using i4l)
;
;msn=39907835
;device => /dev/ttyI0
Index: sip.conf.sample
===================================================================
RCS file: /usr/cvsroot/asterisk/configs/sip.conf.sample,v
retrieving revision 1.27
retrieving revision 1.28
diff -u -d -r1.27 -r1.28
--- sip.conf.sample 8 May 2004 20:58:24 -0000 1.27
+++ sip.conf.sample 24 May 2004 15:09:34 -0000 1.28
@@ -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 @@
;
;register => 1234:password at mysipprovider.com
;
-; Will call to the 's' extension
+; This will pass incoming calls to the 's' extension
;
;
-;register => 2345 at mysipprovider.com/1234
+;register => 2345:password at 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 @@
; 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 at 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 at 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
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