OpenPKG Specs / file revision

 ;

 ; 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@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

 ;

 ;   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)

 				; bindport is the local UDP port that Asterisk will listen on

 ;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

 ;buggymwi=no			; Cisco SIP firmware doesn't support the MWI RFC

 				; fully. Enable this option to not get error messages

 				; when sending MWI to phones with this bug.

 ;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

 ;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. You need to turn this on

 				; in the this section to get any video support at all.

 				; You can turn it off on a per peer basis if the general

 				; video support is enabled, but you can't enable it for

 				; one peer only without enabling in the general section.

 ;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 :-(

 ;matchexterniplocally = yes     ; Only substitute the externip or externhost setting if it matches

                                 ; your localnet setting. Unless you have some sort of strange network

                                 ; setup you will not need to enable this.

 ;

 ; 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

 ;

 ;--------------------------- RTP timers ----------------------------------------------------

 ; These timers are currently used for both audio and video streams. The RTP timeouts

 ; are only applied to the audio channel.

 ; The settings are settable in the global section as well as per device

 ;

 ;rtptimeout=60			; Terminate call if 60 seconds of no RTP or RTCP activity

 				; on the audio channel

 				; when we're not on hold. This is to be able to hangup

 				; a call in the case of a phone disappearing from the net,

 				; like a powerloss or grandma tripping over a cable.

 ;rtpholdtimeout=300		; Terminate call if 300 seconds of no RTP or RTCP activity

 				; on the audio channel

 				; when we're on hold (must be > rtptimeout)

 ;rtpkeepalive=<secs>		; Send keepalives in the RTP stream to keep NAT open

 				; (default is off - zero)

 ;--------------------------- 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 

 ;

 ; You will get more detailed reports (busy etc) if you have a call limit set

 ; for a device. When the call limit is filled, we will indicate busy. Note that

 ; you need at least 2 in order to be able to do attended transfers.

 ;

 ; For queues, you will need this level of detail in status reporting, regardless

 ; if you use SIP subscriptions. Queues and manager use the same internal interface

 ; for reading status information.

 ;

 ; 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 (default: no)

 ;notifyhold = yes		; Notify subscriptions on HOLD state (default: no)

 				; Turning on notifyringing and notifyhold will add a lot

 				; more database transactions if you are using realtime.

 ;limitonpeers = yes		; Apply call limits on peers only. This will improve 

 				; status notification when you are using type=friend

 				; Inbound calls, that really apply to the user part

 				; of a friend will now be added to and compared with

 				; the peer limit instead of applying two call limits,

 				; one for the peer and one for the user.

 				; "sip show inuse" will only show active calls on 

 				; the peer side of a "type=friend" object if this

 				; setting is turned on.

 ;----------------------------------------- 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 

 ; This has to be enabled in the general section for all devices to work. You can then

 ; disable it on a per device basis. 

 ;

 ; T.38 faxing only works in SIP to SIP calls, with no local or agent channel being used.

 ;

 ; 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@mysipprovider.com	

 ;

 ;     This will pass incoming calls to the 's' extension

 ;

 ;

 ;register => 2345:password@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 (;rport)

                                 ; never = Never attempt NAT mode or RFC3581 support

 				; route = Assume NAT, don't send rport 

 				; (work around more UNIDEN bugs)

 ;----------------------------------- MEDIA HANDLING --------------------------------

 ; By default, Asterisk tries to re-invite the audio to an optimal path. If there's

 ; no reason for Asterisk to stay in the media path, the media will be redirected.

 ; This does not really work with in the case where Asterisk is outside and have

 ; clients on the inside of a NAT. In that case, you want to set canreinvite=nonat

 ;

 ;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.

 				; In Asterisk 1.4 this setting also affect direct RTP

 				; at call setup (a new feature in 1.4 - setting up the

 				; call directly between the endpoints instead of sending

 				; a re-INVITE).

 ;directrtpsetup=yes		; Enable the new experimental direct RTP setup. This sets up

 				; the call directly with media peer-2-peer without re-invites.

 				; Will not work for video and cases where the callee sends 

 				; RTP payloads and fmtp headers in the 200 OK that does not match the

 				; callers INVITE. This will also fail if canreinvite is enabled when

 				; the device is actually behind NAT.

 ;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

 ;allowexternaldomains=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@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

 ;port=80				; The port number we want to connect to on the remote side

 					; Also used as "defaultport" in combination with "defaultip" settings

 ;------------------------------------------------------------------------------

 ; 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@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@default,1233@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@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.