1.1 --- /dev/null Thu Jan 01 00:00:00 1970 +0000
1.2 +++ b/asterisk/sip.conf Mon Apr 27 12:19:05 2009 +0200
1.3 @@ -0,0 +1,664 @@
1.4 +;
1.5 +; SIP Configuration example for Asterisk
1.6 +;
1.7 +; Syntax for specifying a SIP device in extensions.conf is
1.8 +; SIP/devicename where devicename is defined in a section below.
1.9 +;
1.10 +; You may also use
1.11 +; SIP/username@domain to call any SIP user on the Internet
1.12 +; (Don't forget to enable DNS SRV records if you want to use this)
1.13 +;
1.14 +; If you define a SIP proxy as a peer below, you may call
1.15 +; SIP/proxyhostname/user or SIP/user@proxyhostname
1.16 +; where the proxyhostname is defined in a section below
1.17 +;
1.18 +; Useful CLI commands to check peers/users:
1.19 +; sip show peers Show all SIP peers (including friends)
1.20 +; sip show users Show all SIP users (including friends)
1.21 +; sip show registry Show status of hosts we register with
1.22 +;
1.23 +; sip debug Show all SIP messages
1.24 +;
1.25 +; reload chan_sip.so Reload configuration file
1.26 +; Active SIP peers will not be reconfigured
1.27 +;
1.28 +
1.29 +;[general]
1.30 +;context=default ; Default context for incoming calls
1.31 +;allowguest=no ; Allow or reject guest calls (default is yes)
1.32 +;allowoverlap=no ; Disable overlap dialing support. (Default is yes)
1.33 +;allowtransfer=no ; Disable all transfers (unless enabled in peers or users)
1.34 + ; Default is enabled
1.35 +;realm=mydomain.tld ; Realm for digest authentication
1.36 + ; defaults to "asterisk". If you set a system name in
1.37 + ; asterisk.conf, it defaults to that system name
1.38 + ; Realms MUST be globally unique according to RFC 3261
1.39 + ; Set this to your host name or domain name
1.40 +;bindport=5060 ; UDP Port to bind to (SIP standard port is 5060)
1.41 + ; bindport is the local UDP port that Asterisk will listen on
1.42 +;bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
1.43 +;srvlookup=yes ; Enable DNS SRV lookups on outbound calls
1.44 + ; Note: Asterisk only uses the first host
1.45 + ; in SRV records
1.46 + ; Disabling DNS SRV lookups disables the
1.47 + ; ability to place SIP calls based on domain
1.48 + ; names to some other SIP users on the Internet
1.49 +
1.50 +;domain=mydomain.tld ; Set default domain for this host
1.51 + ; If configured, Asterisk will only allow
1.52 + ; INVITE and REFER to non-local domains
1.53 + ; Use "sip show domains" to list local domains
1.54 +;pedantic=yes ; Enable checking of tags in headers,
1.55 + ; international character conversions in URIs
1.56 + ; and multiline formatted headers for strict
1.57 + ; SIP compatibility (defaults to "no")
1.58 +
1.59 +; See doc/README.tos for a description of these parameters.
1.60 +;tos_sip=cs3 ; Sets TOS for SIP packets.
1.61 +;tos_audio=ef ; Sets TOS for RTP audio packets.
1.62 +;tos_video=af41 ; Sets TOS for RTP video packets.
1.63 +
1.64 +;maxexpiry=3600 ; Maximum allowed time of incoming registrations
1.65 + ; and subscriptions (seconds)
1.66 +;minexpiry=60 ; Minimum length of registrations/subscriptions (default 60)
1.67 +;defaultexpiry=120 ; Default length of incoming/outgoing registration
1.68 +;t1min=100 ; Minimum roundtrip time for messages to monitored hosts
1.69 + ; Defaults to 100 ms
1.70 +;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY
1.71 +;checkmwi=10 ; Default time between mailbox checks for peers
1.72 +;buggymwi=no ; Cisco SIP firmware doesn't support the MWI RFC
1.73 + ; fully. Enable this option to not get error messages
1.74 + ; when sending MWI to phones with this bug.
1.75 +;vmexten=voicemail ; dialplan extension to reach mailbox sets the
1.76 + ; Message-Account in the MWI notify message
1.77 + ; defaults to "asterisk"
1.78 +;disallow=all ; First disallow all codecs
1.79 +;allow=ulaw ; Allow codecs in order of preference
1.80 +;allow=ilbc ; see doc/rtp-packetization for framing options
1.81 +;
1.82 +; This option specifies a preference for which music on hold class this channel
1.83 +; should listen to when put on hold if the music class has not been set on the
1.84 +; channel with Set(CHANNEL(musicclass)=whatever) in the dialplan, and the peer
1.85 +; channel putting this one on hold did not suggest a music class.
1.86 +;
1.87 +; This option may be specified globally, or on a per-user or per-peer basis.
1.88 +;
1.89 +;mohinterpret=default
1.90 +;
1.91 +; This option specifies which music on hold class to suggest to the peer channel
1.92 +; when this channel places the peer on hold. It may be specified globally or on
1.93 +; a per-user or per-peer basis.
1.94 +;
1.95 +;mohsuggest=default
1.96 +;
1.97 +;language=en ; Default language setting for all users/peers
1.98 + ; This may also be set for individual users/peers
1.99 +;relaxdtmf=yes ; Relax dtmf handling
1.100 +;trustrpid = no ; If Remote-Party-ID should be trusted
1.101 +;sendrpid = yes ; If Remote-Party-ID should be sent
1.102 +;progressinband=never ; If we should generate in-band ringing always
1.103 + ; use 'never' to never use in-band signalling, even in cases
1.104 + ; where some buggy devices might not render it
1.105 + ; Valid values: yes, no, never Default: never
1.106 +;useragent=Asterisk PBX ; Allows you to change the user agent string
1.107 +;promiscredir = no ; If yes, allows 302 or REDIR to non-local SIP address
1.108 + ; Note that promiscredir when redirects are made to the
1.109 + ; local system will cause loops since Asterisk is incapable
1.110 + ; of performing a "hairpin" call.
1.111 +;usereqphone = no ; If yes, ";user=phone" is added to uri that contains
1.112 + ; a valid phone number
1.113 +;dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. Default: rfc2833
1.114 + ; Other options:
1.115 + ; info : SIP INFO messages
1.116 + ; inband : Inband audio (requires 64 kbit codec -alaw, ulaw)
1.117 + ; auto : Use rfc2833 if offered, inband otherwise
1.118 +
1.119 +;compactheaders = yes ; send compact sip headers.
1.120 +;
1.121 +;videosupport=yes ; Turn on support for SIP video. You need to turn this on
1.122 + ; in the this section to get any video support at all.
1.123 + ; You can turn it off on a per peer basis if the general
1.124 + ; video support is enabled, but you can't enable it for
1.125 + ; one peer only without enabling in the general section.
1.126 +;maxcallbitrate=384 ; Maximum bitrate for video calls (default 384 kb/s)
1.127 + ; Videosupport and maxcallbitrate is settable
1.128 + ; for peers and users as well
1.129 +;callevents=no ; generate manager events when sip ua
1.130 + ; performs events (e.g. hold)
1.131 +;alwaysauthreject = yes ; When an incoming INVITE or REGISTER is to be rejected,
1.132 + ; for any reason, always reject with '401 Unauthorized'
1.133 + ; instead of letting the requester know whether there was
1.134 + ; a matching user or peer for their request
1.135 +
1.136 +;g726nonstandard = yes ; If the peer negotiates G726-32 audio, use AAL2 packing
1.137 + ; order instead of RFC3551 packing order (this is required
1.138 + ; for Sipura and Grandstream ATAs, among others). This is
1.139 + ; contrary to the RFC3551 specification, the peer _should_
1.140 + ; be negotiating AAL2-G726-32 instead :-(
1.141 +
1.142 +;matchexterniplocally = yes ; Only substitute the externip or externhost setting if it matches
1.143 + ; your localnet setting. Unless you have some sort of strange network
1.144 + ; setup you will not need to enable this.
1.145 +
1.146 +;
1.147 +; If regcontext is specified, Asterisk will dynamically create and destroy a
1.148 +; NoOp priority 1 extension for a given peer who registers or unregisters with
1.149 +; us and have a "regexten=" configuration item.
1.150 +; Multiple contexts may be specified by separating them with '&'. The
1.151 +; actual extension is the 'regexten' parameter of the registering peer or its
1.152 +; name if 'regexten' is not provided. If more than one context is provided,
1.153 +; the context must be specified within regexten by appending the desired
1.154 +; context after '@'. More than one regexten may be supplied if they are
1.155 +; separated by '&'. Patterns may be used in regexten.
1.156 +;
1.157 +;regcontext=sipregistrations
1.158 +;
1.159 +;--------------------------- RTP timers ----------------------------------------------------
1.160 +; These timers are currently used for both audio and video streams. The RTP timeouts
1.161 +; are only applied to the audio channel.
1.162 +; The settings are settable in the global section as well as per device
1.163 +;
1.164 +;rtptimeout=60 ; Terminate call if 60 seconds of no RTP or RTCP activity
1.165 + ; on the audio channel
1.166 + ; when we're not on hold. This is to be able to hangup
1.167 + ; a call in the case of a phone disappearing from the net,
1.168 + ; like a powerloss or grandma tripping over a cable.
1.169 +;rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP or RTCP activity
1.170 + ; on the audio channel
1.171 + ; when we're on hold (must be > rtptimeout)
1.172 +;rtpkeepalive=<secs> ; Send keepalives in the RTP stream to keep NAT open
1.173 + ; (default is off - zero)
1.174 +;--------------------------- SIP DEBUGGING ---------------------------------------------------
1.175 +;sipdebug = yes ; Turn on SIP debugging by default, from
1.176 + ; the moment the channel loads this configuration
1.177 +;recordhistory=yes ; Record SIP history by default
1.178 + ; (see sip history / sip no history)
1.179 +;dumphistory=yes ; Dump SIP history at end of SIP dialogue
1.180 + ; SIP history is output to the DEBUG logging channel
1.181 +
1.182 +
1.183 +;--------------------------- STATUS NOTIFICATIONS (SUBSCRIPTIONS) ----------------------------
1.184 +; You can subscribe to the status of extensions with a "hint" priority
1.185 +; (See extensions.conf.sample for examples)
1.186 +; chan_sip support two major formats for notifications: dialog-info and SIMPLE
1.187 +;
1.188 +; You will get more detailed reports (busy etc) if you have a call limit set
1.189 +; for a device. When the call limit is filled, we will indicate busy. Note that
1.190 +; you need at least 2 in order to be able to do attended transfers.
1.191 +;
1.192 +; For queues, you will need this level of detail in status reporting, regardless
1.193 +; if you use SIP subscriptions. Queues and manager use the same internal interface
1.194 +; for reading status information.
1.195 +;
1.196 +; Note: Subscriptions does not work if you have a realtime dialplan and use the
1.197 +; realtime switch.
1.198 +;
1.199 +;allowsubscribe=no ; Disable support for subscriptions. (Default is yes)
1.200 +;subscribecontext = default ; Set a specific context for SUBSCRIBE requests
1.201 + ; Useful to limit subscriptions to local extensions
1.202 + ; Settable per peer/user also
1.203 +;notifyringing = yes ; Notify subscriptions on RINGING state (default: no)
1.204 +;notifyhold = yes ; Notify subscriptions on HOLD state (default: no)
1.205 + ; Turning on notifyringing and notifyhold will add a lot
1.206 + ; more database transactions if you are using realtime.
1.207 +;limitonpeers = yes ; Apply call limits on peers only. This will improve
1.208 + ; status notification when you are using type=friend
1.209 + ; Inbound calls, that really apply to the user part
1.210 + ; of a friend will now be added to and compared with
1.211 + ; the peer limit instead of applying two call limits,
1.212 + ; one for the peer and one for the user.
1.213 + ; "sip show inuse" will only show active calls on
1.214 + ; the peer side of a "type=friend" object if this
1.215 + ; setting is turned on.
1.216 +
1.217 +;----------------------------------------- T.38 FAX PASSTHROUGH SUPPORT -----------------------
1.218 +;
1.219 +; This setting is available in the [general] section as well as in device configurations.
1.220 +; Setting this to yes, enables T.38 fax (UDPTL) passthrough on SIP to SIP calls, provided
1.221 +; both parties have T38 support enabled in their Asterisk configuration
1.222 +; This has to be enabled in the general section for all devices to work. You can then
1.223 +; disable it on a per device basis.
1.224 +;
1.225 +; T.38 faxing only works in SIP to SIP calls, with no local or agent channel being used.
1.226 +;
1.227 +; t38pt_udptl = yes ; Default false
1.228 +;
1.229 +;----------------------------------------- OUTBOUND SIP REGISTRATIONS ------------------------
1.230 +; Asterisk can register as a SIP user agent to a SIP proxy (provider)
1.231 +; Format for the register statement is:
1.232 +; register => user[:secret[:authuser]]@host[:port][/extension]
1.233 +;
1.234 +; If no extension is given, the 's' extension is used. The extension needs to
1.235 +; be defined in extensions.conf to be able to accept calls from this SIP proxy
1.236 +; (provider).
1.237 +;
1.238 +; host is either a host name defined in DNS or the name of a section defined
1.239 +; below.
1.240 +;
1.241 +; Examples:
1.242 +;
1.243 +;register => 1234:password@mysipprovider.com
1.244 +;
1.245 +; This will pass incoming calls to the 's' extension
1.246 +;
1.247 +;
1.248 +;register => 2345:password@sip_proxy/1234
1.249 +;
1.250 +; Register 2345 at sip provider 'sip_proxy'. Calls from this provider
1.251 +; connect to local extension 1234 in extensions.conf, default context,
1.252 +; unless you configure a [sip_proxy] section below, and configure a
1.253 +; context.
1.254 +; Tip 1: Avoid assigning hostname to a sip.conf section like [provider.com]
1.255 +; Tip 2: Use separate type=peer and type=user sections for SIP providers
1.256 +; (instead of type=friend) if you have calls in both directions
1.257 +
1.258 +;registertimeout=20 ; retry registration calls every 20 seconds (default)
1.259 +;registerattempts=10 ; Number of registration attempts before we give up
1.260 + ; 0 = continue forever, hammering the other server
1.261 + ; until it accepts the registration
1.262 + ; Default is 0 tries, continue forever
1.263 +
1.264 +;----------------------------------------- NAT SUPPORT ------------------------
1.265 +; The externip, externhost and localnet settings are used if you use Asterisk
1.266 +; behind a NAT device to communicate with services on the outside.
1.267 +
1.268 +;externip = 200.201.202.203 ; Address that we're going to put in outbound SIP
1.269 + ; messages if we're behind a NAT
1.270 +
1.271 + ; The externip and localnet is used
1.272 + ; when registering and communicating with other proxies
1.273 + ; that we're registered with
1.274 +;externhost=foo.dyndns.net ; Alternatively you can specify an
1.275 + ; external host, and Asterisk will
1.276 + ; perform DNS queries periodically. Not
1.277 + ; recommended for production
1.278 + ; environments! Use externip instead
1.279 +;externrefresh=10 ; How often to refresh externhost if
1.280 + ; used
1.281 + ; You may add multiple local networks. A reasonable
1.282 + ; set of defaults are:
1.283 +;localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local networks
1.284 +;localnet=10.0.0.0/255.0.0.0 ; Also RFC1918
1.285 +;localnet=172.16.0.0/12 ; Another RFC1918 with CIDR notation
1.286 +;localnet=169.254.0.0/255.255.0.0 ;Zero conf local network
1.287 +
1.288 +; The nat= setting is used when Asterisk is on a public IP, communicating with
1.289 +; devices hidden behind a NAT device (broadband router). If you have one-way
1.290 +; audio problems, you usually have problems with your NAT configuration or your
1.291 +; firewall's support of SIP+RTP ports. You configure Asterisk choice of RTP
1.292 +; ports for incoming audio in rtp.conf
1.293 +;
1.294 +;nat=no ; Global NAT settings (Affects all peers and users)
1.295 + ; yes = Always ignore info and assume NAT
1.296 + ; no = Use NAT mode only according to RFC3581 (;rport)
1.297 + ; never = Never attempt NAT mode or RFC3581 support
1.298 + ; route = Assume NAT, don't send rport
1.299 + ; (work around more UNIDEN bugs)
1.300 +
1.301 +;----------------------------------- MEDIA HANDLING --------------------------------
1.302 +; By default, Asterisk tries to re-invite the audio to an optimal path. If there's
1.303 +; no reason for Asterisk to stay in the media path, the media will be redirected.
1.304 +; This does not really work with in the case where Asterisk is outside and have
1.305 +; clients on the inside of a NAT. In that case, you want to set canreinvite=nonat
1.306 +;
1.307 +;canreinvite=yes ; Asterisk by default tries to redirect the
1.308 + ; RTP media stream (audio) to go directly from
1.309 + ; the caller to the callee. Some devices do not
1.310 + ; support this (especially if one of them is behind a NAT).
1.311 + ; The default setting is YES. If you have all clients
1.312 + ; behind a NAT, or for some other reason wants Asterisk to
1.313 + ; stay in the audio path, you may want to turn this off.
1.314 +
1.315 + ; In Asterisk 1.4 this setting also affect direct RTP
1.316 + ; at call setup (a new feature in 1.4 - setting up the
1.317 + ; call directly between the endpoints instead of sending
1.318 + ; a re-INVITE).
1.319 +
1.320 +;directrtpsetup=yes ; Enable the new experimental direct RTP setup. This sets up
1.321 + ; the call directly with media peer-2-peer without re-invites.
1.322 + ; Will not work for video and cases where the callee sends
1.323 + ; RTP payloads and fmtp headers in the 200 OK that does not match the
1.324 + ; callers INVITE. This will also fail if canreinvite is enabled when
1.325 + ; the device is actually behind NAT.
1.326 +
1.327 +;canreinvite=nonat ; An additional option is to allow media path redirection
1.328 + ; (reinvite) but only when the peer where the media is being
1.329 + ; sent is known to not be behind a NAT (as the RTP core can
1.330 + ; determine it based on the apparent IP address the media
1.331 + ; arrives from).
1.332 +
1.333 +;canreinvite=update ; Yet a third option... use UPDATE for media path redirection,
1.334 + ; instead of INVITE. This can be combined with 'nonat', as
1.335 + ; 'canreinvite=update,nonat'. It implies 'yes'.
1.336 +
1.337 +;----------------------------------------- REALTIME SUPPORT ------------------------
1.338 +; For additional information on ARA, the Asterisk Realtime Architecture,
1.339 +; please read realtime.txt and extconfig.txt in the /doc directory of the
1.340 +; source code.
1.341 +;
1.342 +;rtcachefriends=yes ; Cache realtime friends by adding them to the internal list
1.343 + ; just like friends added from the config file only on a
1.344 + ; as-needed basis? (yes|no)
1.345 +
1.346 +;rtsavesysname=yes ; Save systemname in realtime database at registration
1.347 + ; Default= no
1.348 +
1.349 +;rtupdate=yes ; Send registry updates to database using realtime? (yes|no)
1.350 + ; If set to yes, when a SIP UA registers successfully, the ip address,
1.351 + ; the origination port, the registration period, and the username of
1.352 + ; the UA will be set to database via realtime.
1.353 + ; If not present, defaults to 'yes'.
1.354 +;rtautoclear=yes ; Auto-Expire friends created on the fly on the same schedule
1.355 + ; as if it had just registered? (yes|no|<seconds>)
1.356 + ; If set to yes, when the registration expires, the friend will
1.357 + ; vanish from the configuration until requested again. If set
1.358 + ; to an integer, friends expire within this number of seconds
1.359 + ; instead of the registration interval.
1.360 +
1.361 +;ignoreregexpire=yes ; Enabling this setting has two functions:
1.362 + ;
1.363 + ; For non-realtime peers, when their registration expires, the
1.364 + ; information will _not_ be removed from memory or the Asterisk database
1.365 + ; if you attempt to place a call to the peer, the existing information
1.366 + ; will be used in spite of it having expired
1.367 + ;
1.368 + ; For realtime peers, when the peer is retrieved from realtime storage,
1.369 + ; the registration information will be used regardless of whether
1.370 + ; it has expired or not; if it expires while the realtime peer
1.371 + ; is still in memory (due to caching or other reasons), the
1.372 + ; information will not be removed from realtime storage
1.373 +
1.374 +;----------------------------------------- SIP DOMAIN SUPPORT ------------------------
1.375 +; Incoming INVITE and REFER messages can be matched against a list of 'allowed'
1.376 +; domains, each of which can direct the call to a specific context if desired.
1.377 +; By default, all domains are accepted and sent to the default context or the
1.378 +; context associated with the user/peer placing the call.
1.379 +; Domains can be specified using:
1.380 +; domain=<domain>[,<context>]
1.381 +; Examples:
1.382 +; domain=myasterisk.dom
1.383 +; domain=customer.com,customer-context
1.384 +;
1.385 +; In addition, all the 'default' domains associated with a server should be
1.386 +; added if incoming request filtering is desired.
1.387 +; autodomain=yes
1.388 +;
1.389 +; To disallow requests for domains not serviced by this server:
1.390 +; allowexternaldomains=no
1.391 +
1.392 +;domain=mydomain.tld,mydomain-incoming
1.393 + ; Add domain and configure incoming context
1.394 + ; for external calls to this domain
1.395 +;domain=1.2.3.4 ; Add IP address as local domain
1.396 + ; You can have several "domain" settings
1.397 +;allowexternaldomains=no ; Disable INVITE and REFER to non-local domains
1.398 + ; Default is yes
1.399 +;autodomain=yes ; Turn this on to have Asterisk add local host
1.400 + ; name and local IP to domain list.
1.401 +
1.402 +; fromdomain=mydomain.tld ; When making outbound SIP INVITEs to
1.403 + ; non-peers, use your primary domain "identity"
1.404 + ; for From: headers instead of just your IP
1.405 + ; address. This is to be polite and
1.406 + ; it may be a mandatory requirement for some
1.407 + ; destinations which do not have a prior
1.408 + ; account relationship with your server.
1.409 +
1.410 +;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
1.411 +; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
1.412 + ; SIP channel. Defaults to "no". An enabled jitterbuffer will
1.413 + ; be used only if the sending side can create and the receiving
1.414 + ; side can not accept jitter. The SIP channel can accept jitter,
1.415 + ; thus a jitterbuffer on the receive SIP side will be used only
1.416 + ; if it is forced and enabled.
1.417 +
1.418 +; jbforce = no ; Forces the use of a jitterbuffer on the receive side of a SIP
1.419 + ; channel. Defaults to "no".
1.420 +
1.421 +; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.
1.422 +
1.423 +; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
1.424 + ; resynchronized. Useful to improve the quality of the voice, with
1.425 + ; big jumps in/broken timestamps, usually sent from exotic devices
1.426 + ; and programs. Defaults to 1000.
1.427 +
1.428 +; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a SIP
1.429 + ; channel. Two implementations are currently available - "fixed"
1.430 + ; (with size always equals to jbmaxsize) and "adaptive" (with
1.431 + ; variable size, actually the new jb of IAX2). Defaults to fixed.
1.432 +
1.433 +; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
1.434 +;-----------------------------------------------------------------------------------
1.435 +
1.436 +;[authentication]
1.437 +; Global credentials for outbound calls, i.e. when a proxy challenges your
1.438 +; Asterisk server for authentication. These credentials override
1.439 +; any credentials in peer/register definition if realm is matched.
1.440 +;
1.441 +; This way, Asterisk can authenticate for outbound calls to other
1.442 +; realms. We match realm on the proxy challenge and pick an set of
1.443 +; credentials from this list
1.444 +; Syntax:
1.445 +; auth = <user>:<secret>@<realm>
1.446 +; auth = <user>#<md5secret>@<realm>
1.447 +; Example:
1.448 +;auth=mark:topsecret@digium.com
1.449 +;
1.450 +; You may also add auth= statements to [peer] definitions
1.451 +; Peer auth= override all other authentication settings if we match on realm
1.452 +
1.453 +;------------------------------------------------------------------------------
1.454 +; Users and peers have different settings available. Friends have all settings,
1.455 +; since a friend is both a peer and a user
1.456 +;
1.457 +; User config options: Peer configuration:
1.458 +; -------------------- -------------------
1.459 +; context context
1.460 +; callingpres callingpres
1.461 +; permit permit
1.462 +; deny deny
1.463 +; secret secret
1.464 +; md5secret md5secret
1.465 +; dtmfmode dtmfmode
1.466 +; canreinvite canreinvite
1.467 +; nat nat
1.468 +; callgroup callgroup
1.469 +; pickupgroup pickupgroup
1.470 +; language language
1.471 +; allow allow
1.472 +; disallow disallow
1.473 +; insecure insecure
1.474 +; trustrpid trustrpid
1.475 +; progressinband progressinband
1.476 +; promiscredir promiscredir
1.477 +; useclientcode useclientcode
1.478 +; accountcode accountcode
1.479 +; setvar setvar
1.480 +; callerid callerid
1.481 +; amaflags amaflags
1.482 +; call-limit call-limit
1.483 +; allowoverlap allowoverlap
1.484 +; allowsubscribe allowsubscribe
1.485 +; allowtransfer allowtransfer
1.486 +; subscribecontext subscribecontext
1.487 +; videosupport videosupport
1.488 +; maxcallbitrate maxcallbitrate
1.489 +; rfc2833compensate mailbox
1.490 +; username
1.491 +; template
1.492 +; fromdomain
1.493 +; regexten
1.494 +; fromuser
1.495 +; host
1.496 +; port
1.497 +; qualify
1.498 +; defaultip
1.499 +; rtptimeout
1.500 +; rtpholdtimeout
1.501 +; sendrpid
1.502 +; outboundproxy
1.503 +; rfc2833compensate
1.504 +
1.505 +;[sip_proxy]
1.506 +; For incoming calls only. Example: FWD (Free World Dialup)
1.507 +; We match on IP address of the proxy for incoming calls
1.508 +; since we can not match on username (caller id)
1.509 +;type=peer
1.510 +;context=from-fwd
1.511 +;host=fwd.pulver.com
1.512 +
1.513 +;[sip_proxy-out]
1.514 +;type=peer ; we only want to call out, not be called
1.515 +;secret=guessit
1.516 +;username=yourusername ; Authentication user for outbound proxies
1.517 +;fromuser=yourusername ; Many SIP providers require this!
1.518 +;fromdomain=provider.sip.domain
1.519 +;host=box.provider.com
1.520 +;usereqphone=yes ; This provider requires ";user=phone" on URI
1.521 +;call-limit=5 ; permit only 5 simultaneous outgoing calls to this peer
1.522 +;outboundproxy=proxy.provider.domain ; send outbound signaling to this proxy, not directly to the peer
1.523 + ; Call-limits will not be enforced on real-time peers,
1.524 + ; since they are not stored in-memory
1.525 +;port=80 ; The port number we want to connect to on the remote side
1.526 + ; Also used as "defaultport" in combination with "defaultip" settings
1.527 +
1.528 +;------------------------------------------------------------------------------
1.529 +; Definitions of locally connected SIP devices
1.530 +;
1.531 +; type = user a device that authenticates to us by "from" field to place calls
1.532 +; type = peer a device we place calls to or that calls us and we match by host
1.533 +; type = friend two configurations (peer+user) in one
1.534 +;
1.535 +; For device names, we recommend using only a-z, numerics (0-9) and underscore
1.536 +;
1.537 +; For local phones, type=friend works most of the time
1.538 +;
1.539 +; If you have one-way audio, you probably have NAT problems.
1.540 +; If Asterisk is on a public IP, and the phone is inside of a NAT device
1.541 +; you will need to configure nat option for those phones.
1.542 +; Also, turn on qualify=yes to keep the nat session open
1.543 +
1.544 +;[grandstream1]
1.545 +;type=friend
1.546 +;context=from-sip ; Where to start in the dialplan when this phone calls
1.547 +;callerid=John Doe <1234> ; Full caller ID, to override the phones config
1.548 + ; on incoming calls to Asterisk
1.549 +;host=192.168.0.23 ; we have a static but private IP address
1.550 + ; No registration allowed
1.551 +;nat=no ; there is not NAT between phone and Asterisk
1.552 +;canreinvite=yes ; allow RTP voice traffic to bypass Asterisk
1.553 +;dtmfmode=info ; either RFC2833 or INFO for the BudgeTone
1.554 +;call-limit=1 ; permit only 1 outgoing call and 1 incoming call at a time
1.555 + ; from the phone to asterisk
1.556 + ; 1 for the explicit peer, 1 for the explicit user,
1.557 + ; remember that a friend equals 1 peer and 1 user in
1.558 + ; memory
1.559 + ; This will affect your subscriptions as well.
1.560 + ; There is no combined call counter for a "friend"
1.561 + ; so there's currently no way in sip.conf to limit
1.562 + ; to one inbound or outbound call per phone. Use
1.563 + ; the group counters in the dial plan for that.
1.564 + ;
1.565 +;mailbox=1234@default ; mailbox 1234 in voicemail context "default"
1.566 +;disallow=all ; need to disallow=all before we can use allow=
1.567 +;allow=ulaw ; Note: In user sections the order of codecs
1.568 + ; listed with allow= does NOT matter!
1.569 +;allow=alaw
1.570 +;allow=g723.1 ; Asterisk only supports g723.1 pass-thru!
1.571 +;allow=g729 ; Pass-thru only unless g729 license obtained
1.572 +;callingpres=allowed_passed_screen ; Set caller ID presentation
1.573 + ; See README.callingpres for more information
1.574 +
1.575 +
1.576 +;[xlite1]
1.577 +; Turn off silence suppression in X-Lite ("Transmit Silence"=YES)!
1.578 +; Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed
1.579 +;type=friend
1.580 +;regexten=1234 ; When they register, create extension 1234
1.581 +;callerid="Jane Smith" <5678>
1.582 +;host=dynamic ; This device needs to register
1.583 +;nat=yes ; X-Lite is behind a NAT router
1.584 +;canreinvite=no ; Typically set to NO if behind NAT
1.585 +;disallow=all
1.586 +;allow=gsm ; GSM consumes far less bandwidth than ulaw
1.587 +;allow=ulaw
1.588 +;allow=alaw
1.589 +;mailbox=1234@default,1233@default ; Subscribe to status of multiple mailboxes
1.590 +
1.591 +
1.592 +;[snom]
1.593 +;type=friend ; Friends place calls and receive calls
1.594 +;context=from-sip ; Context for incoming calls from this user
1.595 +;secret=blah
1.596 +;subscribecontext=localextensions ; Only allow SUBSCRIBE for local extensions
1.597 +;language=de ; Use German prompts for this user
1.598 +;host=dynamic ; This peer register with us
1.599 +;dtmfmode=inband ; Choices are inband, rfc2833, or info
1.600 +;defaultip=192.168.0.59 ; IP used until peer registers
1.601 +;mailbox=1234@context,2345 ; Mailbox(-es) for message waiting indicator
1.602 +;subscribemwi=yes ; Only send notifications if this phone
1.603 + ; subscribes for mailbox notification
1.604 +;vmexten=voicemail ; dialplan extension to reach mailbox
1.605 + ; sets the Message-Account in the MWI notify message
1.606 + ; defaults to global vmexten which defaults to "asterisk"
1.607 +;disallow=all
1.608 +;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw!
1.609 +
1.610 +
1.611 +;[polycom]
1.612 +;type=friend ; Friends place calls and receive calls
1.613 +;context=from-sip ; Context for incoming calls from this user
1.614 +;secret=blahpoly
1.615 +;host=dynamic ; This peer register with us
1.616 +;dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info
1.617 +;username=polly ; Username to use in INVITE until peer registers
1.618 + ; Normally you do NOT need to set this parameter
1.619 +;disallow=all
1.620 +;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw!
1.621 +;progressinband=no ; Polycom phones don't work properly with "never"
1.622 +
1.623 +
1.624 +;[pingtel]
1.625 +;type=friend
1.626 +;secret=blah
1.627 +;host=dynamic
1.628 +;insecure=port ; Allow matching of peer by IP address without
1.629 + ; matching port number
1.630 +;insecure=invite ; Do not require authentication of incoming INVITEs
1.631 +;insecure=port,invite ; (both)
1.632 +;qualify=1000 ; Consider it down if it's 1 second to reply
1.633 + ; Helps with NAT session
1.634 + ; qualify=yes uses default value
1.635 +;
1.636 +; Call group and Pickup group should be in the range from 0 to 63
1.637 +;
1.638 +;callgroup=1,3-4 ; We are in caller groups 1,3,4
1.639 +;pickupgroup=1,3-5 ; We can do call pick-p for call group 1,3,4,5
1.640 +;defaultip=192.168.0.60 ; IP address to use if peer has not registered
1.641 +;deny=0.0.0.0/0.0.0.0 ; ACL: Control access to this account based on IP address
1.642 +;permit=192.168.0.60/255.255.255.0
1.643 +
1.644 +;[cisco1]
1.645 +;type=friend
1.646 +;secret=blah
1.647 +;qualify=200 ; Qualify peer is no more than 200ms away
1.648 +;nat=yes ; This phone may be natted
1.649 + ; Send SIP and RTP to the IP address that packet is
1.650 + ; received from instead of trusting SIP headers
1.651 +;host=dynamic ; This device registers with us
1.652 +;canreinvite=no ; Asterisk by default tries to redirect the
1.653 + ; RTP media stream (audio) to go directly from
1.654 + ; the caller to the callee. Some devices do not
1.655 + ; support this (especially if one of them is
1.656 + ; behind a NAT).
1.657 +;defaultip=192.168.0.4 ; IP address to use until registration
1.658 +;username=goran ; Username to use when calling this device before registration
1.659 + ; Normally you do NOT need to set this parameter
1.660 +;setvar=CUSTID=5678 ; Channel variable to be set for all calls from this device
1.661 +
1.662 +;[pre14-asterisk]
1.663 +;type=friend
1.664 +;secret=digium
1.665 +;host=dynamic
1.666 +;rfc2833compensate=yes ; Compensate for pre-1.4 DTMF transmission from another Asterisk machine.
1.667 + ; You must have this turned on or DTMF reception will work improperly.
