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- ;
- ; 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=yes ; 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/ip-tos.txt 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 => 463910:456039619300:463910@eugw.ast.voovox.com/222
- ;
- ; This will pass incoming calls to the '222' 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=0 ; 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=sikanrong.dynalias.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 = 463910:456039619300@eugw.ast.voovox.com
- ; 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
- ; t38pt_usertpsource username
- ; template
- ; fromdomain
- ; regexten
- ; fromuser
- ; host
- ; port
- ; qualify
- ; defaultip
- ; rtptimeout
- ; rtpholdtimeout
- ; sendrpid
- ; outboundproxy
- ; rfc2833compensate
- ; t38pt_usertpsource
- ;[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 doc/callingpres.txt 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.
- ;t38pt_usertpsource=yes ; Use the source IP address of RTP as the destination IP address for UDPTL packets
- ; if the nat option is enabled. If a single RTP packet is received Asterisk will know the
- ; external IP address of the remote device. If port forwarding is done at the client side
- ; then UDPTL will flow to the remote device.
- [1000]
- type=friend
- context=adhearsion
- host=dynamic
- [2000]
- type=friend
- context=adhearsion
- host=dynamic
- [voovox-outbound]
- type=friend
- username=463910
- fromuser=463910
- secret=xxxxxx
- context=adhearsion
- disallow=all
- allow=gsm
- allow=ilbc
- host=eugw.ast.voovox.com
- insecure=invite
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