mirror of https://github.com/asterisk/asterisk
You can not select more than 25 topics
Topics must start with a letter or number, can include dashes ('-') and can be up to 35 characters long.
1118 lines
60 KiB
1118 lines
60 KiB
;
|
|
; SIP Configuration example for Asterisk
|
|
;
|
|
; SIP dial strings
|
|
;-----------------------------------------------------------
|
|
; In the dialplan (extensions.conf) you can use several
|
|
; syntaxes for dialing SIP devices.
|
|
; SIP/devicename
|
|
; SIP/username@domain (SIP uri)
|
|
; SIP/username[:password[:md5secret[:authname[:transport]]]]@host[:port]
|
|
; SIP/devicename/extension
|
|
;
|
|
;
|
|
; Devicename
|
|
; devicename is defined as a peer in a section below.
|
|
;
|
|
; username@domain
|
|
; Call any SIP user on the Internet
|
|
; (Don't forget to enable DNS SRV records if you want to use this)
|
|
;
|
|
; devicename/extension
|
|
; 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
|
|
; This syntax also works with ATA's with FXO ports
|
|
;
|
|
; SIP/username[:password[:md5secret[:authname]]]@host[:port]
|
|
; This form allows you to specify password or md5secret and authname
|
|
; without altering any authentication data in config.
|
|
; Examples:
|
|
;
|
|
; SIP/*98@mysipproxy
|
|
; SIP/sales:topsecret::account02@domain.com:5062
|
|
; SIP/12345678::bc53f0ba8ceb1ded2b70e05c3f91de4f:myname@192.168.0.1
|
|
;
|
|
; All of these dial strings specify the SIP request URI.
|
|
; In addition, you can specify a specific To: header by adding an
|
|
; exclamation mark after the dial string, like
|
|
;
|
|
; SIP/sales@mysipproxy!sales@edvina.net
|
|
;
|
|
; CLI Commands
|
|
; -------------------------------------------------------------
|
|
; Useful CLI commands to check peers/users:
|
|
; sip show peers Show all SIP peers (including friends)
|
|
; sip show registry Show status of hosts we register with
|
|
;
|
|
; sip set debug on Show all SIP messages
|
|
;
|
|
; module reload chan_sip.so Reload configuration file
|
|
;
|
|
;------- Naming devices ------------------------------------------------------
|
|
;
|
|
; When naming devices, make sure you understand how Asterisk matches calls
|
|
; that come in.
|
|
; 1. Asterisk checks the SIP From: address username and matches against
|
|
; names of devices with type=user
|
|
; The name is the text between square brackets [name]
|
|
; 2. Asterisk checks the From: addres and matches the list of devices
|
|
; with a type=peer
|
|
; 3. Asterisk checks the IP address (and port number) that the INVITE
|
|
; was sent from and matches against any devices with type=peer
|
|
;
|
|
; Don't mix extensions with the names of the devices. Devices need a unique
|
|
; name. The device name is *not* used as phone numbers. Phone numbers are
|
|
; anything you declare as an extension in the dialplan (extensions.conf).
|
|
;
|
|
; When setting up trunks, make sure there's no risk that any From: username
|
|
; (caller ID) will match any of your device names, because then Asterisk
|
|
; might match the wrong device.
|
|
;
|
|
; Note: The parameter "username" is not the username and in most cases is
|
|
; not needed at all. Check below. In later releases, it's renamed
|
|
; to "defaultuser" which is a better name, since it is used in
|
|
; combination with the "defaultip" setting.
|
|
;-----------------------------------------------------------------------------
|
|
|
|
; ** Deprecated configuration options **
|
|
; The "call-limit" configuation option is deprecated. It still works in
|
|
; this version of Asterisk, but will disappear in the next version.
|
|
; You are encouraged to use the dialplan groupcount functionality
|
|
; to enforce call limits instead of using this channel-specific method.
|
|
;
|
|
; You can still set limits per device in sip.conf or in a database by using
|
|
; "setvar" to set variables that can be used in the dialplan for various limits.
|
|
|
|
[general]
|
|
context=default ; Default context for incoming calls
|
|
;allowguest=no ; Allow or reject guest calls (default is yes)
|
|
;match_auth_username=yes ; if available, match user entry using the
|
|
; 'username' field from the authentication line
|
|
; instead of the From: field.
|
|
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
|
|
udpbindaddr=0.0.0.0 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all)
|
|
; Optionally add a port number, 192.168.1.1:5062 (default is port 5060)
|
|
|
|
;
|
|
; Note that the TCP and TLS support for chan_sip is currently considered
|
|
; experimental. Since it is new, all of the related configuration options are
|
|
; subject to change in any release. If they are changed, the changes will
|
|
; be reflected in this sample configuration file, as well as in the UPGRADE.txt file.
|
|
;
|
|
tcpenable=no ; Enable server for incoming TCP connections (default is no)
|
|
tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces)
|
|
; Optionally add a port number, 192.168.1.1:5062 (default is port 5060)
|
|
|
|
;tlsenable=no ; Enable server for incoming TLS (secure) connections (default is no)
|
|
;tlsbindaddr=0.0.0.0 ; IP address for TLS server to bind to (0.0.0.0) binds to all interfaces)
|
|
; Optionally add a port number, 192.168.1.1:5063 (default is port 5061)
|
|
; Remember that the IP address must match the common name (hostname) in the
|
|
; certificate, so you don't want to bind a TLS socket to multiple IP addresses.
|
|
|
|
;tlscertfile=</path/to/certificate.pem> ; Certificate file (*.pem only) to use for TLS connections
|
|
; default is to look for "asterisk.pem" in current directory
|
|
|
|
;tlsprivatekey=</path/to/private.pem> ; Private key file (*.pem only) for TLS connections.
|
|
; If no tlsprivatekey is specified, tlscertfile is searched for
|
|
; for both public and private key.
|
|
|
|
;tlscafile=</path/to/certificate>
|
|
; If the server your connecting to uses a self signed certificate
|
|
; you should have their certificate installed here so the code can
|
|
; verify the authenticity of their certificate.
|
|
|
|
;tlscadir=</path/to/ca/dir>
|
|
; A directory full of CA certificates. The files must be named with
|
|
; the CA subject name hash value.
|
|
; (see man SSL_CTX_load_verify_locations for more info)
|
|
|
|
;tlsdontverifyserver=[yes|no]
|
|
; If set to yes, don't verify the servers certificate when acting as
|
|
; a client. If you don't have the server's CA certificate you can
|
|
; set this and it will connect without requiring tlscafile to be set.
|
|
; Default is no.
|
|
|
|
;tlscipher=<SSL cipher string>
|
|
; A string specifying which SSL ciphers to use or not use
|
|
; A list of valid SSL cipher strings can be found at:
|
|
; http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS
|
|
;
|
|
;tlsclientmethod=tlsv1 ; values include tlsv1, sslv3, sslv2.
|
|
; Specify protocol for outbound client connections.
|
|
; If left unspecified, the default is sslv2.
|
|
|
|
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
|
|
|
|
;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 qos.tex or Quality of Service section of asterisk.pdf 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.
|
|
;tos_text=af41 ; Sets TOS for RTP text packets.
|
|
|
|
;cos_sip=3 ; Sets 802.1p priority for SIP packets.
|
|
;cos_audio=5 ; Sets 802.1p priority for RTP audio packets.
|
|
;cos_video=4 ; Sets 802.1p priority for RTP video packets.
|
|
;cos_text=3 ; Sets 802.1p priority for RTP text 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
|
|
;mwiexpiry=3600 ; Expiry time for outgoing MWI subscriptions
|
|
;qualifyfreq=60 ; Qualification: How often to check for the
|
|
; host to be up in seconds
|
|
; Set to low value if you use low timeout for
|
|
; NAT of UDP sessions
|
|
;qualifygap=100 ; Number of milliseconds between each group of peers being qualified
|
|
;qualifypeers=1 ; Number of peers in a group to be qualified at the same time
|
|
;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY
|
|
;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.
|
|
;mwi_from=asterisk ; When sending MWI NOTIFY requests, use this setting in
|
|
; the From: header as the "name" portion. Also fill the
|
|
; "user" portion of the URI in the From: header with this
|
|
; value if no fromuser is set
|
|
; Default: empty
|
|
;vmexten=voicemail ; dialplan extension to reach mailbox sets the
|
|
; Message-Account in the MWI notify message
|
|
; defaults to "asterisk"
|
|
|
|
;preferred_codec_only=yes ; Respond to a SIP invite with the single most preferred codec
|
|
; rather than advertising all joint codec capabilities. This
|
|
; limits the other side's codec choice to exactly what we prefer.
|
|
|
|
;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
|
|
;
|
|
;parkinglot=plaza ; Sets the default parking lot for call parking
|
|
; This may also be set for individual users/peers
|
|
; Parkinglots are configured in features.conf
|
|
;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
|
|
;sendrpid = rpid ; Use the "Remote-Party-ID" header
|
|
; to send the identity of the remote party
|
|
; This is identical to sendrpid=yes
|
|
;sendrpid = pai ; Use the "P-Asserted-Identity" header
|
|
; to send the identity of the remote party
|
|
;rpid_update = no ; In certain cases, the only method by which a connected line
|
|
; change may be immediately transmitted is with a SIP UPDATE request.
|
|
; If communicating with another Asterisk server, and you wish to be able
|
|
; transmit such UPDATE messages to it, then you must enable this option.
|
|
; Otherwise, we will have to wait until we can send a reinvite to
|
|
; transmit the information.
|
|
|
|
;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
|
|
; The default user agent string also contains the Asterisk
|
|
; version. If you don't want to expose this, change the
|
|
; useragent string.
|
|
;sdpsession=Asterisk PBX ; Allows you to change the SDP session name string, (s=)
|
|
; Like the useragent parameter, the default user agent string
|
|
; also contains the Asterisk version.
|
|
;sdpowner=root ; Allows you to change the username field in the SDP owner string, (o=)
|
|
; This field MUST NOT contain spaces
|
|
;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 (application/dtmf-relay)
|
|
; shortinfo : SIP INFO messages (application/dtmf)
|
|
; 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 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.
|
|
; If you set videosupport to "always", then RTP ports will
|
|
; always be set up for video, even on clients that don't
|
|
; support it. This assists callfile-derived calls and
|
|
; certain transferred calls to use always use video when
|
|
; available. [yes|NO|always]
|
|
|
|
;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)
|
|
;authfailureevents=no ; generate manager "peerstatus" events when peer can't
|
|
; authenticate with Asterisk. Peerstatus will be "rejected".
|
|
;alwaysauthreject = yes ; When an incoming INVITE or REGISTER is to be rejected,
|
|
; for any reason, always reject with an identical response
|
|
; equivalent to valid username and invalid password/hash
|
|
; instead of letting the requester know whether there was
|
|
; a matching user or peer for their request. This reduces
|
|
; the ability of an attacker to scan for valid SIP usernames.
|
|
|
|
;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 :-(
|
|
;outboundproxy=proxy.provider.domain ; send outbound signaling to this proxy, not directly to the devices
|
|
;outboundproxy=proxy.provider.domain:8080 ; send outbound signaling to this proxy, not directly to the devices
|
|
;outboundproxy=proxy.provider.domain,force ; Send ALL outbound signalling to proxy, ignoring route: headers
|
|
;outboundproxy=tls://proxy.provider.domain ; same as '=proxy.provider.domain' except we try to connect with tls
|
|
; ; (could also be tcp,udp) - defining transports on the proxy line only
|
|
; ; applies for the global proxy, otherwise use the transport= option
|
|
;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.
|
|
|
|
;dynamic_exclude_static = yes ; Disallow all dynamic hosts from registering
|
|
; as any IP address used for staticly defined
|
|
; hosts. This helps avoid the configuration
|
|
; error of allowing your users to register at
|
|
; the same address as a SIP provider.
|
|
|
|
;contactdeny=0.0.0.0/0.0.0.0 ; Use contactpermit and contactdeny to
|
|
;contactpermit=172.16.0.0/255.255.0.0 ; restrict at what IPs your users may
|
|
; register their phones.
|
|
|
|
;engine=asterisk ; RTP engine to use when communicating with the device
|
|
|
|
;
|
|
; 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
|
|
;regextenonqualify=yes ; Default "no"
|
|
; If you have qualify on and the peer becomes unreachable
|
|
; this setting will enforce inactivation of the regexten
|
|
; extension for the peer
|
|
;
|
|
;--------------------------- SIP timers ----------------------------------------------------
|
|
; These timers are used primarily in INVITE transactions.
|
|
; The default for Timer T1 is 500 ms or the measured run-trip time between
|
|
; Asterisk and the device if you have qualify=yes for the device.
|
|
;
|
|
;t1min=100 ; Minimum roundtrip time for messages to monitored hosts
|
|
; Defaults to 100 ms
|
|
;timert1=500 ; Default T1 timer
|
|
; Defaults to 500 ms or the measured round-trip
|
|
; time to a peer (qualify=yes).
|
|
;timerb=32000 ; Call setup timer. If a provisional response is not received
|
|
; in this amount of time, the call will autocongest
|
|
; Defaults to 64*timert1
|
|
|
|
;--------------------------- 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 Session-Timers (RFC 4028)------------------------------------
|
|
; SIP Session-Timers provide an end-to-end keep-alive mechanism for active SIP sessions.
|
|
; This mechanism can detect and reclaim SIP channels that do not terminate through normal
|
|
; signaling procedures. Session-Timers can be configured globally or at a user/peer level.
|
|
; The operation of Session-Timers is driven by the following configuration parameters:
|
|
;
|
|
; * session-timers - Session-Timers feature operates in the following three modes:
|
|
; originate : Request and run session-timers always
|
|
; accept : Run session-timers only when requested by other UA
|
|
; refuse : Do not run session timers in any case
|
|
; The default mode of operation is 'accept'.
|
|
; * session-expires - Maximum session refresh interval in seconds. Defaults to 1800 secs.
|
|
; * session-minse - Minimum session refresh interval in seconds. Defualts to 90 secs.
|
|
; * session-refresher - The session refresher (uac|uas). Defaults to 'uas'.
|
|
;
|
|
;session-timers=originate
|
|
;session-expires=600
|
|
;session-minse=90
|
|
;session-refresher=uas
|
|
;
|
|
;--------------------------- HASH TABLE SIZES ------------------------------------------------
|
|
; For maximum efficiency, adjust the following
|
|
; values to be slightly larger than the maximum number of in-memory objects (devices).
|
|
; Too large, and space is wasted. Too small, and things will run slower.
|
|
; 563 is probably way too big for small (home) applications, but it
|
|
; should cover most small/medium sites.
|
|
; It is recommended to make the sizes be a prime number!
|
|
; This was internally set to 17 for small-memory applications...
|
|
; All tables default to 563, except when compiled in LOW_MEMORY mode,
|
|
; in which case, they default to 17. You can override this by uncommenting
|
|
; the following, and changing the values.
|
|
;hash_users=563
|
|
;hash_peers=563
|
|
;hash_dialogs=563
|
|
|
|
;--------------------------- 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 counter enabled
|
|
; for a device.
|
|
;
|
|
; If you set the busylevel, we will indicate busy when we have a number of calls that
|
|
; matches the busylevel treshold.
|
|
;
|
|
; 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 = no ; Control whether subscriptions already INUSE get sent
|
|
; RINGING when another call is sent (default: yes)
|
|
;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.
|
|
;notifycid = yes ; Control whether caller ID information is sent along with
|
|
; dialog-info+xml notifications (supported by snom phones).
|
|
; Note that this feature will only work properly when the
|
|
; incoming call is using the same extension and context that
|
|
; is being used as the hint for the called extension. This means
|
|
; that it won't work when using subscribecontext for your sip
|
|
; user or peer (if subscribecontext is different than context).
|
|
; This is also limited to a single caller, meaning that if an
|
|
; extension is ringing because multiple calls are incoming,
|
|
; only one will be used as the source of caller ID. Specify
|
|
; 'ignore-context' to ignore the called context when looking
|
|
; for the caller's channel. The default value is 'no.' Setting
|
|
; notifycid to 'ignore-context' also causes call-pickups attempted
|
|
; via SNOM's NOTIFY mechanism to set the context for the call pickup
|
|
; to PICKUPMARK.
|
|
;callcounter = yes ; Enable call counters on devices. This can be set per
|
|
; device too.
|
|
|
|
;----------------------------------------- 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. It defaults to off.
|
|
;
|
|
; t38pt_udptl = yes ; Enables T.38 with FEC error correction.
|
|
; t38pt_udptl = yes,fec ; Enables T.38 with FEC error correction.
|
|
; t38pt_udptl = yes,redundancy ; Enables T.38 with redundancy error correction.
|
|
; t38pt_udptl = yes,none ; Enables T.38 with no error correction.
|
|
;
|
|
; Faxs Detect will cause the SIP channel to jump to the 'fax' extension (if it exists)
|
|
; after T.38 is successfully negotiated.
|
|
;
|
|
; faxdetect = 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 => [transport://]user[@domain][:secret[:authuser]]@host[:port][/extension][~expiry]
|
|
;
|
|
;
|
|
;
|
|
; domain is either
|
|
; - domain in DNS
|
|
; - host name in DNS
|
|
; - the name of a peer defined below or in realtime
|
|
; The domain is where you register your username, so your SIP uri you are registering to
|
|
; is username@domain
|
|
;
|
|
; 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).
|
|
;
|
|
; A similar effect can be achieved by adding a "callbackextension" option in a peer section.
|
|
; this is equivalent to having the following line in the general section:
|
|
;
|
|
; register => username:secret@host/callbackextension
|
|
;
|
|
; and more readable because you don't have to write the parameters in two places
|
|
; (note that the "port" is ignored - this is a bug that should be fixed).
|
|
;
|
|
; Note that a register= line doesn't mean that we will match the incoming call in any
|
|
; other way than described above. If you want to control where the call enters your
|
|
; dialplan, which context, you want to define a peer with the hostname of the provider's
|
|
; server. If the provider has multiple servers to place calls to your system, you need
|
|
; a peer for each server.
|
|
;
|
|
; Beginning with Asterisk version 1.6.2, the "user" portion of the register line may
|
|
; contain a port number. Since the logical separator between a host and port number is a
|
|
; ':' character, and this character is already used to separate between the optional "secret"
|
|
; and "authuser" portions of the line, there is a bit of a hoop to jump through if you wish
|
|
; to use a port here. That is, you must explicitly provide a "secret" and "authuser" even if
|
|
; they are blank. See the third example below for an illustration.
|
|
;
|
|
;
|
|
; Examples:
|
|
;
|
|
;register => 1234:password@mysipprovider.com
|
|
;
|
|
; This will pass incoming calls to the 's' extension
|
|
;
|
|
;
|
|
;register => 2345:password@sip_proxy/1234
|
|
;
|
|
; Register 2345 at sip provider 'sip_proxy'. Calls from this provider
|
|
; connect to local extension 1234 in extensions.conf, default context,
|
|
; unless you configure a [sip_proxy] section below, and configure a
|
|
; context.
|
|
; Tip 1: Avoid assigning hostname to a sip.conf section like [provider.com]
|
|
; Tip 2: Use separate inbound and outbound sections for SIP providers
|
|
; (instead of type=friend) if you have calls in both directions
|
|
;
|
|
;register => 3456@mydomain:5082::@mysipprovider.com
|
|
;
|
|
; Note that in this example, the optional authuser and secret portions have
|
|
; been left blank because we have specified a port in the user section
|
|
|
|
;registertimeout=20 ; retry registration calls every 20 seconds (default)
|
|
;registerattempts=10 ; Number of registration attempts before we give up
|
|
; 0 = continue forever, hammering the other server
|
|
; until it accepts the registration
|
|
; Default is 0 tries, continue forever
|
|
;----------------------------------------- OUTBOUND MWI SUBSCRIPTIONS -------------------------
|
|
; Asterisk can subscribe to receive the MWI from another SIP server and store it locally for retrieval
|
|
; by other phones.
|
|
; Format for the mwi register statement is:
|
|
; mwi => user[:secret[:authuser]]@host[:port][/mailbox]
|
|
;
|
|
; Examples:
|
|
;mwi => 1234:password@mysipprovider.com/1234
|
|
;
|
|
; MWI received will be stored in the 1234 mailbox of the SIP_Remote context. It can be used by other phones by following the below:
|
|
; mailbox=1234@SIP_Remote
|
|
;----------------------------------------- NAT SUPPORT ------------------------
|
|
;
|
|
; WARNING: SIP operation behind a NAT is tricky and you really need
|
|
; to read and understand well the following section.
|
|
;
|
|
; When Asterisk is behind a NAT device, the "local" address (and port) that
|
|
; a socket is bound to has different values when seen from the inside or
|
|
; from the outside of the NATted network. Unfortunately this address must
|
|
; be communicated to the outside (e.g. in SIP and SDP messages), and in
|
|
; order to determine the correct value Asterisk needs to know:
|
|
;
|
|
; + whether it is talking to someone "inside" or "outside" of the NATted network.
|
|
; This is configured by assigning the "localnet" parameter with a list
|
|
; of network addresses that are considered "inside" of the NATted network.
|
|
; IF LOCALNET IS NOT SET, THE EXTERNAL ADDRESS WILL NOT BE SET CORRECTLY.
|
|
; Multiple entries are allowed, e.g. a reasonable set is the following:
|
|
;
|
|
; localnet=192.168.0.0/255.255.0.0 ; RFC 1918 addresses
|
|
; 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 "externally visible" address and port number to be used when talking
|
|
; to a host outside the NAT. This information is derived by one of the
|
|
; following (mutually exclusive) config file parameters:
|
|
;
|
|
; a. "externip = hostname[:port]" specifies a static address[:port] to
|
|
; be used in SIP and SDP messages.
|
|
; The hostname is looked up only once, when [re]loading sip.conf .
|
|
; If a port number is not present, use the "bindport" value (which is
|
|
; not guaranteed to work correctly, because a NAT box might remap the
|
|
; port number as well as the address).
|
|
; This approach can be useful if you have a NAT device where you can
|
|
; configure the mapping statically. Examples:
|
|
;
|
|
; externip = 12.34.56.78 ; use this address.
|
|
; externip = 12.34.56.78:9900 ; use this address and port.
|
|
; externip = mynat.my.org:12600 ; Public address of my nat box.
|
|
;
|
|
; b. "externhost = hostname[:port]" is similar to "externip" except
|
|
; that the hostname is looked up every "externrefresh" seconds
|
|
; (default 10s). This can be useful when your NAT device lets you choose
|
|
; the port mapping, but the IP address is dynamic.
|
|
; Beware, you might suffer from service disruption when the name server
|
|
; resolution fails. Examples:
|
|
;
|
|
; externhost=foo.dyndns.net ; refreshed periodically
|
|
; externrefresh=180 ; change the refresh interval
|
|
;
|
|
; c. "stunaddr = stun.server[:port]" queries the STUN server specified
|
|
; as an argument to obtain the external address/port.
|
|
; Queries are also sent periodically every "externrefresh" seconds
|
|
; (as a side effect, sending the query also acts as a keepalive for
|
|
; the state entry on the nat box):
|
|
;
|
|
; stunaddr = foo.stun.com:3478
|
|
; externrefresh = 15
|
|
;
|
|
; Note that at the moment all these mechanism work only for the SIP socket.
|
|
; The IP address discovered with externip/externhost/STUN is reused for
|
|
; media sessions as well, but the port numbers are not remapped so you
|
|
; may still experience problems.
|
|
;
|
|
; NOTE 1: in some cases, NAT boxes will use different port numbers in
|
|
; the internal<->external mapping. In these cases, the "externip" and
|
|
; "externhost" might not help you configure addresses properly, and you
|
|
; really need to use STUN.
|
|
;
|
|
; NOTE 2: when using "externip" or "externhost", the address part is
|
|
; also used as the external address for media sessions.
|
|
; If you use "stunaddr", STUN queries will be sent to the same server
|
|
; also from media sockets, and this should permit a correct mapping of
|
|
; the port numbers as well.
|
|
;
|
|
; In addition to the above, Asterisk has an additional "nat" parameter to
|
|
; address NAT-related issues in incoming SIP or media sessions.
|
|
; In particular, depending on the 'nat= ' settings described below, Asterisk
|
|
; may override the address/port information specified in the SIP/SDP messages,
|
|
; and use the information (sender address) supplied by the network stack instead.
|
|
; However, this is only useful if the external traffic can reach us.
|
|
; The following settings are allowed (both globally and in individual sections):
|
|
;
|
|
; nat = no ; Default. Use rport if the remote side says to use it.
|
|
; nat = force_rport ; Force rport to always be on.
|
|
; nat = yes ; Force rport to always be on and perform symmetric RTP.
|
|
; nat = comedia ; Use rport if the remote side says to use it and perform symmetric RTP.
|
|
|
|
;----------------------------------- 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.
|
|
|
|
; 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'.
|
|
|
|
;ignoresdpversion=yes ; By default, Asterisk will honor the session version
|
|
; number in SDP packets and will only modify the SDP
|
|
; session if the version number changes. This option will
|
|
; force asterisk to ignore the SDP session version number
|
|
; and treat all SDP data as new data. This is required
|
|
; for devices that send us non standard SDP packets
|
|
; (observed with Microsoft OCS). By default this option is
|
|
; off.
|
|
|
|
;----------------------------------------- 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'. Note: realtime peers will
|
|
; probably not function across reloads in the way that you expect, if
|
|
; you turn this option off.
|
|
;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.
|
|
; REGISTER to non-local domains will be automatically denied if a domain
|
|
; list is configured.
|
|
;
|
|
; Domains can be specified using:
|
|
; domain=<domain>[,<context>]
|
|
; Examples:
|
|
; domain=myasterisk.dom
|
|
; domain=customer.com,customer-context
|
|
;
|
|
; In addition, all the 'default' domains associated with a server should be
|
|
; added if incoming request filtering is desired.
|
|
; autodomain=yes
|
|
;
|
|
; To disallow requests for domains not serviced by this server:
|
|
; allowexternaldomains=no
|
|
|
|
;domain=mydomain.tld,mydomain-incoming
|
|
; Add domain and configure incoming context
|
|
; for external calls to this domain
|
|
;domain=1.2.3.4 ; Add IP address as local domain
|
|
; You can have several "domain" settings
|
|
;allowexternaldomains=no ; Disable INVITE and REFER to non-local domains
|
|
; Default is yes
|
|
;autodomain=yes ; Turn this on to have Asterisk add local host
|
|
; name and local IP to domain list.
|
|
|
|
; fromdomain=mydomain.tld ; When making outbound SIP INVITEs to
|
|
; non-peers, use your primary domain "identity"
|
|
; for From: headers instead of just your IP
|
|
; address. This is to be polite and
|
|
; it may be a mandatory requirement for some
|
|
; destinations which do not have a prior
|
|
; account relationship with your server.
|
|
|
|
;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
|
|
; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
|
|
; SIP channel. Defaults to "no". An enabled jitterbuffer will
|
|
; be used only if the sending side can create and the receiving
|
|
; side can not accept jitter. The SIP channel can accept jitter,
|
|
; thus a jitterbuffer on the receive SIP side will be used only
|
|
; if it is forced and enabled.
|
|
|
|
; jbforce = no ; Forces the use of a jitterbuffer on the receive side of a SIP
|
|
; channel. Defaults to "no".
|
|
|
|
; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.
|
|
|
|
; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
|
|
; resynchronized. Useful to improve the quality of the voice, with
|
|
; big jumps in/broken timestamps, usually sent from exotic devices
|
|
; and programs. Defaults to 1000.
|
|
|
|
; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a SIP
|
|
; channel. Two implementations are currently available - "fixed"
|
|
; (with size always equals to jbmaxsize) and "adaptive" (with
|
|
; variable size, actually the new jb of IAX2). Defaults to fixed.
|
|
|
|
; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
|
|
;-----------------------------------------------------------------------------------
|
|
|
|
[authentication]
|
|
; Global credentials for outbound calls, i.e. when a proxy challenges your
|
|
; Asterisk server for authentication. These credentials override
|
|
; any credentials in peer/register definition if realm is matched.
|
|
;
|
|
; This way, Asterisk can authenticate for outbound calls to other
|
|
; realms. We match realm on the proxy challenge and pick an set of
|
|
; credentials from this list
|
|
; Syntax:
|
|
; auth = <user>:<secret>@<realm>
|
|
; auth = <user>#<md5secret>@<realm>
|
|
; Example:
|
|
;auth=mark:topsecret@digium.com
|
|
;
|
|
; You may also add auth= statements to [peer] definitions
|
|
; Peer auth= override all other authentication settings if we match on realm
|
|
|
|
;------------------------------------------------------------------------------
|
|
; DEVICE CONFIGURATION
|
|
;
|
|
; The SIP channel has two types of devices, the friend and the peer.
|
|
; * The type=friend is a device type that accepts both incoming and outbound calls,
|
|
; where Asterisk match on the From: username on incoming calls.
|
|
; (A synonym for friend is "user"). This is a type you use for your local
|
|
; SIP phones.
|
|
; * The type=peer also handles both incoming and outbound calls. On inbound calls,
|
|
; Asterisk only matches on IP/port, not on names. This is mostly used for SIP
|
|
; trunks.
|
|
;
|
|
; 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
|
|
;
|
|
; Configuration options available
|
|
; --------------------
|
|
; context
|
|
; callingpres
|
|
; permit
|
|
; deny
|
|
; secret
|
|
; md5secret
|
|
; remotesecret
|
|
; transport
|
|
; dtmfmode
|
|
; canreinvite
|
|
; nat
|
|
; callgroup
|
|
; pickupgroup
|
|
; language
|
|
; allow
|
|
; disallow
|
|
; insecure
|
|
; trustrpid
|
|
; progressinband
|
|
; promiscredir
|
|
; useclientcode
|
|
; accountcode
|
|
; setvar
|
|
; callerid
|
|
; amaflags
|
|
; callcounter
|
|
; busylevel
|
|
; allowoverlap
|
|
; allowsubscribe
|
|
; allowtransfer
|
|
; ignoresdpversion
|
|
; subscribecontext
|
|
; template
|
|
; videosupport
|
|
; maxcallbitrate
|
|
; rfc2833compensate
|
|
; mailbox
|
|
; session-timers
|
|
; session-expires
|
|
; session-minse
|
|
; session-refresher
|
|
; t38pt_usertpsource
|
|
; regexten
|
|
; fromdomain
|
|
; fromuser
|
|
; host
|
|
; port
|
|
; qualify
|
|
; defaultip
|
|
; defaultuser
|
|
; rtptimeout
|
|
; rtpholdtimeout
|
|
; sendrpid
|
|
; outboundproxy
|
|
; rfc2833compensate
|
|
; callbackextension
|
|
; registertrying
|
|
; timert1
|
|
; timerb
|
|
; qualifyfreq
|
|
; t38pt_usertpsource
|
|
; contactpermit ; Limit what a host may register as (a neat trick
|
|
; contactdeny ; is to register at the same IP as a SIP provider,
|
|
; ; then call oneself, and get redirected to that
|
|
; ; same location).
|
|
|
|
;[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
|
|
;remotesecret=guessit ; Our password to their service
|
|
;defaultuser=yourusername ; Authentication user for outbound proxies
|
|
;fromuser=yourusername ; Many SIP providers require this!
|
|
;fromdomain=provider.sip.domain
|
|
;host=box.provider.com
|
|
;transport=udp,tcp ; This sets the default transport type to udp for outgoing, and will
|
|
; ; accept both tcp and udp. The default transport type is only used for
|
|
; ; outbound messages until a Registration takes place. During the
|
|
; ; peer Registration the transport type may change to another supported
|
|
; ; type if the peer requests so.
|
|
|
|
;usereqphone=yes ; This provider requires ";user=phone" on URI
|
|
;callcounter=yes ; Enable call counter
|
|
;busylevel=2 ; Signal busy at 2 or more calls
|
|
;outboundproxy=proxy.provider.domain ; send outbound signaling to this proxy, not directly to the peer
|
|
;port=80 ; The port number we want to connect to on the remote side
|
|
; Also used as "defaultport" in combination with "defaultip" settings
|
|
|
|
;--- sample definition for a provider
|
|
;[provider1]
|
|
;type=peer
|
|
;host=sip.provider1.com
|
|
;fromuser=4015552299 ; how your provider knows you
|
|
;remotesecret=youwillneverguessit ; The password we use to authenticate to them
|
|
;secret=gissadetdu ; The password they use to contact us
|
|
;callbackextension=123 ; Register with this server and require calls coming back to this extension
|
|
;transport=udp,tcp ; This sets the transport type to udp for outgoing, and will
|
|
; ; accept both tcp and udp. Default is udp. The first transport
|
|
; ; listed will always be used for outgoing connections.
|
|
|
|
;
|
|
; Because you might have a large number of similar sections, it is generally
|
|
; convenient to use templates for the common parameters, and add them
|
|
; the the various sections. Examples are below, and we can even leave
|
|
; the templates uncommented as they will not harm:
|
|
|
|
[basic-options](!) ; a template
|
|
dtmfmode=rfc2833
|
|
context=from-office
|
|
type=friend
|
|
|
|
[natted-phone](!,basic-options) ; another template inheriting basic-options
|
|
nat=yes
|
|
canreinvite=no
|
|
host=dynamic
|
|
|
|
[public-phone](!,basic-options) ; another template inheriting basic-options
|
|
nat=no
|
|
canreinvite=yes
|
|
|
|
[my-codecs](!) ; a template for my preferred codecs
|
|
disallow=all
|
|
allow=ilbc
|
|
allow=g729
|
|
allow=gsm
|
|
allow=g723
|
|
allow=ulaw
|
|
|
|
[ulaw-phone](!) ; and another one for ulaw-only
|
|
disallow=all
|
|
allow=ulaw
|
|
|
|
; and finally instantiate a few phones
|
|
;
|
|
; [2133](natted-phone,my-codecs)
|
|
; secret = peekaboo
|
|
; [2134](natted-phone,ulaw-phone)
|
|
; secret = not_very_secret
|
|
; [2136](public-phone,ulaw-phone)
|
|
; secret = not_very_secret_either
|
|
; ...
|
|
;
|
|
|
|
; Standard configurations not using templates look like this:
|
|
;
|
|
;[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 (deprecated)
|
|
; 1 for the explicit peer, 1 for the explicit user,
|
|
; remember that a friend equals 1 peer and 1 user in
|
|
; memory
|
|
; There is no combined call counter for a "friend"
|
|
; so there's currently no way in sip.conf to limit
|
|
; to one inbound or outbound call per phone. Use
|
|
; the group counters in the dial plan for that.
|
|
;
|
|
;mailbox=1234@default ; mailbox 1234 in voicemail context "default"
|
|
;disallow=all ; need to disallow=all before we can use allow=
|
|
;allow=ulaw ; Note: In user sections the order of codecs
|
|
; listed with allow= does NOT matter!
|
|
;allow=alaw
|
|
;allow=g723.1 ; Asterisk only supports g723.1 pass-thru!
|
|
;allow=g729 ; Pass-thru only unless g729 license obtained
|
|
;callingpres=allowed_passed_screen ; Set caller ID presentation
|
|
; See README.callingpres for more information
|
|
|
|
;[xlite1]
|
|
; Turn off silence suppression in X-Lite ("Transmit Silence"=YES)!
|
|
; Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed
|
|
;type=friend
|
|
;regexten=1234 ; When they register, create extension 1234
|
|
;callerid="Jane Smith" <5678>
|
|
;host=dynamic ; This device needs to register
|
|
;nat=yes ; X-Lite is behind a NAT router
|
|
;canreinvite=no ; Typically set to NO if behind NAT
|
|
;disallow=all
|
|
;allow=gsm ; GSM consumes far less bandwidth than ulaw
|
|
;allow=ulaw
|
|
;allow=alaw
|
|
;mailbox=1234@default,1233@default ; Subscribe to status of multiple mailboxes
|
|
;registertrying=yes ; Send a 100 Trying when the device registers.
|
|
|
|
;[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
|
|
;defaultuser=polly ; Username to use in INVITE until peer registers
|
|
;defaultip=192.168.40.123
|
|
; 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
|
|
;qualifyfreq=60 ; Qualification: How often to check for the
|
|
; host to be up in seconds
|
|
; Set to low value if you use low timeout for
|
|
; NAT of UDP sessions
|
|
;
|
|
; 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
|
|
;defaultuser=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 or to this device
|
|
;setvar=ATTENDED_TRANSFER_COMPLETE_SOUND=beep ; This channel variable will
|
|
; cause the given audio file to
|
|
; be played upon completion of
|
|
; an attended transfer.
|
|
|
|
;[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.
|