[asterisk-commits] oej: branch oej/codename-pineapple r44695 - /team/oej/codename-pineapple/conf...

asterisk-commits at lists.digium.com asterisk-commits at lists.digium.com
Sat Oct 7 08:31:33 MST 2006


Author: oej
Date: Sat Oct  7 10:31:32 2006
New Revision: 44695

URL: http://svn.digium.com/view/asterisk?rev=44695&view=rev
Log:
Configuration...

Added:
    team/oej/codename-pineapple/configs/sip3.conf.sample   (with props)

Added: team/oej/codename-pineapple/configs/sip3.conf.sample
URL: http://svn.digium.com/view/asterisk/team/oej/codename-pineapple/configs/sip3.conf.sample?rev=44695&view=auto
==============================================================================
--- team/oej/codename-pineapple/configs/sip3.conf.sample (added)
+++ team/oej/codename-pineapple/configs/sip3.conf.sample Sat Oct  7 10:31:32 2006
@@ -1,0 +1,596 @@
+;
+; SIP Configuration example 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 at 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 at proxyhostname 
+; where the proxyhostname is defined in a section below 
+; 
+; Useful CLI commands to check peers/users:
+;   sip list peers		Show all SIP peers (including friends)
+;   sip list users		Show all SIP users (including friends)
+;   sip list registry		Show status of hosts we register with
+;
+;   sip debug			Show all SIP messages
+;
+;   module reload chan_sip.so	Reload configuration file
+;				Active SIP peers will not be reconfigured
+;
+
+[general]
+context=default			; Default context for incoming calls
+;allowguest=no			; Allow or reject guest calls (default is yes)
+allowoverlap=no			; Disable overlap dialing support. (Default is yes)
+;allowtransfer=no		; Disable all transfers (unless enabled in peers or users)
+				; Default is enabled
+;realm=mydomain.tld		; Realm for digest authentication
+				; defaults to "asterisk". If you set a system name in
+				; asterisk.conf, it defaults to that system name
+				; Realms MUST be globally unique according to RFC 3261
+				; Set this to your host name or domain name
+bindport=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
+				; Disabling DNS SRV lookups disables the 
+				; ability to place SIP calls based on domain 
+				; names to some other SIP users on the Internet
+				
+;domain=mydomain.tld		; Set default domain for this host
+				; If configured, Asterisk will only allow
+				; INVITE and REFER to non-local domains
+				; Use "sip show domains" to list local domains
+;pedantic=yes			; Enable checking of tags in headers, 
+				; international character conversions in URIs
+				; and multiline formatted headers for strict
+				; SIP compatibility (defaults to "no")
+
+; See doc/README.tos for a description of these parameters.
+;tos_sip=cs3                    ; Sets TOS for SIP packets.
+;tos_audio=ef                   ; Sets TOS for RTP audio packets.
+;tos_video=af41                 ; Sets TOS for RTP video packets.
+
+;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/outgoing registration
+;t1min=100			; Minimum roundtrip time for messages to monitored hosts
+				; Defaults to 100 ms
+;notifymimetype=text/plain	; Allow overriding of mime type in MWI NOTIFY
+;checkmwi=10			; Default time between mailbox checks for peers
+;vmexten=voicemail		; dialplan extension to reach mailbox sets the 
+				; Message-Account in the MWI notify message 
+				; defaults to "asterisk"
+;disallow=all			; First disallow all codecs
+;allow=ulaw			; Allow codecs in order of preference
+;allow=ilbc			; see doc/rtp-packetization for framing options
+;
+; This option specifies a preference for which music on hold class this channel
+; should listen to when put on hold if the music class has not been set on the
+; channel with Set(CHANNEL(musicclass)=whatever) in the dialplan, and the peer
+; channel putting this one on hold did not suggest a music class.
+;
+; This option may be specified globally, or on a per-user or per-peer basis.
+;
+;mohinterpret=default
+;
+; This option specifies which music on hold class to suggest to the peer channel
+; when this channel places the peer on hold. It may be specified globally or on
+; a per-user or per-peer basis.
+;
+;mohsuggest=default
+;
+;language=en			; Default language setting for all users/peers
+				; This may also be set for individual users/peers
+;relaxdtmf=yes			; Relax dtmf handling
+;rtptimeout=60			; Terminate call if 60 seconds of no RTP activity
+				; when we're not on hold
+;rtpholdtimeout=300		; Terminate call if 300 seconds of no RTP activity
+				; when we're on hold (must be > rtptimeout)
+;trustrpid = no			; If Remote-Party-ID should be trusted
+;sendrpid = yes			; If Remote-Party-ID should be sent
+;progressinband=never		; If we should generate in-band ringing always
+				; use 'never' to never use in-band signalling, even in cases
+				; where some buggy devices might not render it
+				; Valid values: yes, no, never Default: never
+;useragent=Asterisk PBX		; Allows you to change the user agent string
+;promiscredir = no      	; If yes, allows 302 or REDIR to non-local SIP address
+	                       	; Note that promiscredir when redirects are made to the
+       	                	; local system will cause loops since Asterisk is incapable
+       	                	; of performing a "hairpin" call.
+;usereqphone = no		; If yes, ";user=phone" is added to uri that contains
+				; a valid phone number
+;dtmfmode = rfc2833		; Set default dtmfmode for sending DTMF. Default: rfc2833
+				; Other options: 
+				; info : SIP INFO messages
+				; inband : Inband audio (requires 64 kbit codec -alaw, ulaw)
+				; auto : Use rfc2833 if offered, inband otherwise
+
+;compactheaders = yes		; send compact sip headers.
+;
+;videosupport=yes		; Turn on support for SIP video
+;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)
+;alwaysauthreject = yes		; When an incoming INVITE or REGISTER is to be rejected,
+ 		    		; for any reason, always reject with '401 Unauthorized'
+ 				; instead of letting the requester know whether there was
+ 				; a matching user or peer for their request
+
+;g726nonstandard = yes		; If the peer negotiates G726-32 audio, use AAL2 packing
+				; order instead of RFC3551 packing order (this is required
+				; for Sipura and Grandstream ATAs, among others). This is
+				; contrary to the RFC3551 specification, the peer _should_
+				; be negotiating AAL2-G726-32 instead :-(
+
+;
+; If regcontext is specified, Asterisk will dynamically create and destroy a
+; NoOp priority 1 extension for a given peer who registers or unregisters with
+; us and have a "regexten=" configuration item.  
+; Multiple contexts may be specified by separating them with '&'. The 
+; actual extension is the 'regexten' parameter of the registering peer or its
+; name if 'regexten' is not provided.  If more than one context is provided,
+; the context must be specified within regexten by appending the desired
+; context after '@'.  More than one regexten may be supplied if they are 
+; separated by '&'.  Patterns may be used in regexten.
+;
+;regcontext=sipregistrations
+;
+;--------------------------- SIP DEBUGGING ---------------------------------------------------
+;sipdebug = yes			; Turn on SIP debugging by default, from
+				; the moment the channel loads this configuration
+;recordhistory=yes		; Record SIP history by default 
+				; (see sip history / sip no history)
+;dumphistory=yes		; Dump SIP history at end of SIP dialogue
+				; SIP history is output to the DEBUG logging channel
+
+
+;--------------------------- STATUS NOTIFICATIONS (SUBSCRIPTIONS) ----------------------------
+; You can subscribe to the status of extensions with a "hint" priority
+; (See extensions.conf.sample for examples)
+; chan_sip support two major formats for notifications: dialog-info and SIMPLE 
+; Note: Subscriptions does not work if you have a realtime dialplan and use the
+; realtime switch.
+;
+;allowsubscribe=no		; Disable support for subscriptions. (Default is yes)
+;subscribecontext = default	; Set a specific context for SUBSCRIBE requests
+				; Useful to limit subscriptions to local extensions
+				; Settable per peer/user also
+;notifyringing = yes		; Notify subscriptions on RINGING state
+;----------------------------------------- T.38 FAX PASSTHROUGH SUPPORT -----------------------
+;
+; This setting is available in the [general] section as well as in device configurations.
+; Setting this to yes, enables T.38 fax (UDPTL) passthrough on SIP to SIP calls, provided
+; both parties have T38 support enabled in their Asterisk configuration (either general or
+; peer/user/friend sections)
+;
+; t38pt_udptl = yes            ; Default false
+;
+;----------------------------------------- OUTBOUND SIP REGISTRATIONS  ------------------------
+; Asterisk can register as a SIP user agent to a SIP proxy (provider)
+; Format for the register statement is:
+;       register => user[:secret[:authuser]]@host[:port][/extension]
+;
+; If no extension is given, the 's' extension is used. The extension needs to
+; be defined in extensions.conf to be able to accept calls from this SIP proxy
+; (provider).
+;
+; host is either a host name defined in DNS or the name of a section defined
+; below.
+;
+; Examples:
+;
+;register => 1234:password at mysipprovider.com	
+;
+;     This will pass incoming calls to the 's' extension
+;
+;
+;register => 2345:password at sip_proxy/1234
+;
+;    Register 2345 at sip provider 'sip_proxy'.  Calls from this provider
+;    connect to local extension 1234 in extensions.conf, default 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
+  
+;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
+				; 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
+
+				; The externip and localnet is used
+				; when registering and communicating with other proxies
+				; that we're registered with
+;externhost=foo.dyndns.net	; Alternatively you can specify an 
+				; external host, and Asterisk will 
+				; perform DNS queries periodically.  Not
+				; recommended for production 
+				; 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:
+;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
+
+; The nat= setting is used when Asterisk is on a public IP, communicating with
+; devices hidden behind a NAT device (broadband router).  If you have one-way
+; audio problems, you usually have problems with your NAT configuration or your
+; firewall's support of SIP+RTP ports.  You configure Asterisk choice of RTP
+; ports for incoming audio in rtp.conf
+;
+;nat=no				; Global NAT settings  (Affects all peers and users)
+                                ; yes = Always ignore info and assume NAT
+                                ; no = Use NAT mode only according to RFC3581 
+                                ; never = Never attempt NAT mode or RFC3581 support
+				; route = Assume NAT, don't send rport 
+				; (work around more UNIDEN bugs)
+
+;canreinvite=yes		; 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).
+				; The default setting is YES. If you have all clients
+				; behind a NAT, or for some other reason wants Asterisk to
+				; stay in the audio path, you may want to turn this off.
+
+;canreinvite=nonat		; An additional option is to allow media path redirection
+				; (reinvite) but only when the peer where the media is being
+				; sent is known to not be behind a NAT (as the RTP core can
+				; determine it based on the apparent IP address the media
+				; arrives from).
+
+;canreinvite=update		; Yet a third option... use UPDATE for media path redirection,
+				; instead of INVITE. This can be combined with 'nonat', as
+				; 'canreinvite=update,nonat'. It implies 'yes'.
+
+;----------------------------------------- REALTIME SUPPORT ------------------------
+; For additional information on ARA, the Asterisk Realtime Architecture,
+; please read realtime.txt and extconfig.txt in the /doc directory of the
+; source code.
+;
+;rtcachefriends=yes		; Cache realtime friends by adding them to the internal list
+				; just like friends added from the config file only on a
+				; as-needed basis? (yes|no)
+
+;rtsavesysname=yes		; Save systemname in realtime database at registration
+				; Default= no
+
+;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'.
+;rtautoclear=yes		; Auto-Expire friends created on the fly on the same schedule
+				; as if it had just registered? (yes|no|<seconds>)
+				; 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 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
+
+;----------------------------------------- SIP DOMAIN SUPPORT ------------------------
+; Incoming INVITE and REFER messages can be matched against a list of 'allowed'
+; domains, each of which can direct the call to a specific context if desired.
+; By default, all domains are accepted and sent to the default context or the
+; context associated with the user/peer placing the call.
+; Domains can be specified using:
+; domain=<domain>[,<context>]
+; Examples:
+; domain=myasterisk.dom
+; domain=customer.com,customer-context
+;
+; In addition, all the 'default' domains associated with a server should be
+; added if incoming request filtering is desired.
+; autodomain=yes
+;
+; To disallow requests for domains not serviced by this server:
+; allowexternaldomains=no
+
+;domain=mydomain.tld,mydomain-incoming
+				; Add domain and configure incoming context
+				; for external calls to this domain
+;domain=1.2.3.4			; Add IP address as local domain
+				; You can have several "domain" settings
+;allowexternalinvites=no	; Disable INVITE and REFER to non-local domains
+				; Default is yes
+;autodomain=yes			; Turn this on to have Asterisk add local host
+				; name and local IP to domain list.
+
+; fromdomain=mydomain.tld 	; When making outbound SIP INVITEs to
+                          	; non-peers, use your primary domain "identity"
+                          	; for From: headers instead of just your IP
+                          	; address. This is to be polite and
+                          	; it may be a mandatory requirement for some
+                          	; destinations which do not have a prior
+                          	; account relationship with your server. 
+
+;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
+; jbenable = yes              ; Enables the use of a jitterbuffer on the receiving side of a
+                              ; SIP channel. Defaults to "no". An enabled jitterbuffer will
+                              ; be used only if the sending side can create and the receiving
+                              ; side can not accept jitter. The SIP channel can accept jitter,
+                              ; thus a jitterbuffer on the receive SIP side will be used only
+                              ; if it is forced and enabled.
+
+; jbforce = no                ; Forces the use of a jitterbuffer on the receive side of a SIP
+                              ; channel. Defaults to "no".
+
+; jbmaxsize = 200             ; Max length of the jitterbuffer in milliseconds.
+
+; jbresyncthreshold = 1000    ; Jump in the frame timestamps over which the jitterbuffer is
+                              ; resynchronized. Useful to improve the quality of the voice, with
+                              ; big jumps in/broken timestamps, usually sent from exotic devices
+                              ; and programs. Defaults to 1000.
+
+; jbimpl = fixed              ; Jitterbuffer implementation, used on the receiving side of a SIP
+                              ; channel. Two implementations are currently available - "fixed"
+                              ; (with size always equals to jbmaxsize) and "adaptive" (with
+                              ; variable size, actually the new jb of IAX2). Defaults to fixed.
+
+; jblog = no                  ; Enables jitterbuffer frame logging. Defaults to "no".
+;-----------------------------------------------------------------------------------
+
+[authentication]
+; Global credentials for outbound calls, i.e. when a proxy challenges your
+; Asterisk server for authentication. These credentials override
+; any credentials in peer/register definition if realm is matched.
+;
+; This way, Asterisk can authenticate for outbound calls to other
+; realms. We match realm on the proxy challenge and pick an set of 
+; credentials from this list
+; Syntax:
+;	auth = <user>:<secret>@<realm>
+;	auth = <user>#<md5secret>@<realm>
+; Example:
+;auth=mark:topsecret at digium.com
+; 
+; You may also add auth= statements to [peer] definitions 
+; Peer auth= override all other authentication settings if we match on realm
+
+;------------------------------------------------------------------------------
+; 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
+; callingpres		      callingpres
+; permit                      permit
+; deny                        deny
+; secret                      secret
+; md5secret                   md5secret
+; dtmfmode                    dtmfmode
+; canreinvite                 canreinvite
+; nat                         nat
+; callgroup                   callgroup
+; pickupgroup                 pickupgroup
+; language                    language
+; allow                       allow
+; disallow                    disallow
+; insecure                    insecure
+; trustrpid                   trustrpid
+; progressinband              progressinband
+; promiscredir                promiscredir
+; useclientcode               useclientcode
+; accountcode                 accountcode
+; setvar                      setvar
+; callerid		      callerid
+; amaflags		      amaflags
+; call-limit		      call-limit
+; allowoverlap		      allowoverlap
+; allowsubscribe	      allowsubscribe
+; allowtransfer	      	      allowtransfer
+; subscribecontext	      subscribecontext
+; videosupport		      videosupport
+; maxcallbitrate	      maxcallbitrate
+; rfc2833compensate           mailbox
+;                             username
+;                             template
+;                             fromdomain
+;                             regexten
+;                             fromuser
+;                             host
+;                             port
+;                             qualify
+;                             defaultip
+;                             rtptimeout
+;                             rtpholdtimeout
+;                             sendrpid
+;                             outboundproxy
+;                             rfc2833compensate
+
+;[sip_proxy]
+; For incoming calls only. Example: FWD (Free World Dialup)
+; We match on IP address of the proxy for incoming calls 
+; since we can not match on username (caller id)
+;type=peer
+;context=from-fwd
+;host=fwd.pulver.com
+
+;[sip_proxy-out]
+;type=peer          			; we only want to call out, not be called
+;secret=guessit
+;username=yourusername			; Authentication user for outbound proxies
+;fromuser=yourusername			; Many SIP providers require this!
+;fromdomain=provider.sip.domain	
+;host=box.provider.com
+;usereqphone=yes			; This provider requires ";user=phone" on URI
+;call-limit=5				; permit only 5 simultaneous outgoing calls to this peer
+;outboundproxy=proxy.provider.domain	; send outbound signaling to this proxy, not directly to the peer
+				; Call-limits will not be enforced on real-time peers,
+				; since they are not stored in-memory
+
+;------------------------------------------------------------------------------
+; 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 probably have NAT problems. 
+; If Asterisk is on a public IP, and the phone is inside of a NAT device
+; you will need to configure nat option for those phones.
+; Also, turn on qualify=yes to keep the nat session open
+
+;[grandstream1]
+;type=friend 			
+;context=from-sip		; Where to start in the dialplan when this phone calls
+;callerid=John Doe <1234>	; Full caller ID, to override the phones config
+				; on incoming calls to Asterisk
+;host=192.168.0.23		; we have a static but private IP address
+				; No registration allowed
+;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
+;call-limit=1			; permit only 1 outgoing call and 1 incoming call at a time
+				; from the phone to asterisk
+				; 1 for the explicit peer, 1 for the explicit user,
+				; remember that a friend equals 1 peer and 1 user in
+				; memory
+				; This will affect your subscriptions as well.
+				; There is no combined call counter for a "friend"
+				; so there's currently no way in sip.conf to limit
+				; to one inbound or outbound call per phone. Use
+				; the group counters in the dial plan for that.
+				;
+;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
+;callingpres=allowed_passed_screen	; Set caller ID presentation
+				; See README.callingpres for more information
+
+
+;[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
+;regexten=1234			; When they register, create extension 1234
+;callerid="Jane Smith" <5678>
+;host=dynamic			; This device needs to register
+;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
+;mailbox=1234 at default,1233 at default	; Subscribe to status of multiple mailboxes
+
+
+;[snom]
+;type=friend			; Friends place calls and receive calls
+;context=from-sip		; Context for incoming calls from this user
+;secret=blah
+;subscribecontext=localextensions	; Only allow SUBSCRIBE for local extensions
+;language=de			; Use German prompts for this user 
+;host=dynamic			; This peer register with us
+;dtmfmode=inband		; Choices are inband, rfc2833, or info
+;defaultip=192.168.0.59		; IP used until peer registers
+;mailbox=1234 at context,2345      ; Mailbox(-es) for message waiting indicator
+;subscribemwi=yes		; Only send notifications if this phone 
+				; subscribes for mailbox notification
+;vmexten=voicemail		; dialplan extension to reach mailbox 
+				; sets the Message-Account in the MWI notify message
+				; defaults to global vmexten which defaults to "asterisk"
+;disallow=all
+;allow=ulaw			; dtmfmode=inband only works with ulaw or alaw!
+
+
+;[polycom]
+;type=friend			; Friends place calls and receive calls
+;context=from-sip		; Context for incoming calls from this user
+;secret=blahpoly
+;host=dynamic			; This peer register with us
+;dtmfmode=rfc2833		; Choices are inband, rfc2833, or info
+;username=polly			; Username to use in INVITE until peer registers
+				; Normally you do NOT need to set this parameter
+;disallow=all
+;allow=ulaw                     ; dtmfmode=inband only works with ulaw or alaw!
+;progressinband=no		; Polycom phones don't work properly with "never"
+
+
+;[pingtel]
+;type=friend
+;secret=blah
+;host=dynamic
+;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
+				; Helps with NAT session
+				; qualify=yes uses default value
+;
+; Call group and Pickup group should be in the range from 0 to 63
+;
+;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 registered
+;deny=0.0.0.0/0.0.0.0		; ACL: Control access to this account based on IP address
+;permit=192.168.0.60/255.255.255.0
+
+;[cisco1]
+;type=friend
+;secret=blah
+;qualify=200			; Qualify peer is no more than 200ms away
+;nat=yes			; This phone may be natted
+				; Send SIP and RTP to the 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).
+;defaultip=192.168.0.4		; IP address to use until registration
+;username=goran			; Username to use when calling this device before registration
+				; Normally you do NOT need to set this parameter
+;setvar=CUSTID=5678		; Channel variable to be set for all calls from this device
+
+;[pre14-asterisk]
+;type=friend
+;secret=digium
+;host=dynamic
+;rfc2833compensate=yes		; Compensate for pre-1.4 DTMF transmission from another Asterisk machine.
+				; You must have this turned on or DTMF reception will work improperly.

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