1.1 --- a/asterisk/sip.conf Sun Mar 20 19:27:35 2011 +0100
1.2 +++ /dev/null Thu Jan 01 00:00:00 1970 +0000
1.3 @@ -1,664 +0,0 @@
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.
