1556 lines
89 KiB
Plaintext
1556 lines
89 KiB
Plaintext
;
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; SIP Configuration example for Asterisk
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;
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; Note: Please read the security documentation for Asterisk in order to
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; understand the risks of installing Asterisk with the sample
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; configuration. If your Asterisk is installed on a public
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; IP address connected to the Internet, you will want to learn
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; about the various security settings BEFORE you start
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; Asterisk.
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;
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; Especially note the following settings:
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; - allowguest (default enabled)
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; - permit/deny/acl - IP address filters
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; - contactpermit/contactdeny/contactacl - IP address filters for registrations
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; - context - Which set of services you offer various users
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;
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; SIP dial strings
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;-----------------------------------------------------------
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; In the dialplan (extensions.conf) you can use several
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; syntaxes for dialing SIP devices.
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; SIP/devicename
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; SIP/username@domain (SIP uri)
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; SIP/username[:password[:md5secret[:authname[:transport]]]]@host[:port]
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; SIP/devicename/extension
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; SIP/devicename/extension/IPorHost
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; SIP/username@domain//IPorHost
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;
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;
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; Devicename
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; devicename is defined as a peer in a section below.
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;
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; username@domain
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; Call any SIP user on the Internet
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; (Don't forget to enable DNS SRV records if you want to use this)
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;
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; devicename/extension
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; If you define a SIP proxy as a peer below, you may call
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; SIP/proxyhostname/user or SIP/user@proxyhostname
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; where the proxyhostname is defined in a section below
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; This syntax also works with ATA's with FXO ports
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;
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; SIP/username[:password[:md5secret[:authname]]]@host[:port]
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; This form allows you to specify password or md5secret and authname
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; without altering any authentication data in config.
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; Examples:
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;
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; SIP/*98@mysipproxy
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; SIP/sales:topsecret::account02@domain.com:5062
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; SIP/12345678::bc53f0ba8ceb1ded2b70e05c3f91de4f:myname@192.168.0.1
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;
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; IPorHost
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; The next server for this call regardless of domain/peer
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;
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; All of these dial strings specify the SIP request URI.
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; In addition, you can specify a specific To: header by adding an
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; exclamation mark after the dial string, like
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;
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; SIP/sales@mysipproxy!sales@edvina.net
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;
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; A new feature for 1.8 allows one to specify a host or IP address to use
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; when routing the call. This is typically used in tandem with func_srv if
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; multiple methods of reaching the same domain exist. The host or IP address
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; is specified after the third slash in the dialstring. Examples:
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;
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; SIP/devicename/extension/IPorHost
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; SIP/username@domain//IPorHost
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;
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; CLI Commands
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; -------------------------------------------------------------
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; Useful CLI commands to check peers/users:
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; sip show peers Show all SIP peers (including friends)
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; sip show registry Show status of hosts we register with
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;
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; sip set debug on Show all SIP messages
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;
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; sip reload Reload configuration file
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; sip show settings Show the current channel configuration
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;
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;------- Naming devices ------------------------------------------------------
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;
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; When naming devices, make sure you understand how Asterisk matches calls
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; that come in.
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; 1. Asterisk checks the SIP From: address username and matches against
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; names of devices with type=user
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; The name is the text between square brackets [name]
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; 2. Asterisk checks the From: addres and matches the list of devices
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; with a type=peer
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; 3. Asterisk checks the IP address (and port number) that the INVITE
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; was sent from and matches against any devices with type=peer
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;
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; Don't mix extensions with the names of the devices. Devices need a unique
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; name. The device name is *not* used as phone numbers. Phone numbers are
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; anything you declare as an extension in the dialplan (extensions.conf).
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;
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; When setting up trunks, make sure there's no risk that any From: username
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; (caller ID) will match any of your device names, because then Asterisk
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; might match the wrong device.
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;
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; Note: The parameter "username" is not the username and in most cases is
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; not needed at all. Check below. In later releases, it's renamed
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; to "defaultuser" which is a better name, since it is used in
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; combination with the "defaultip" setting.
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;-----------------------------------------------------------------------------
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; ** Old configuration options **
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; The "call-limit" configuation option is considered old is replaced
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; by new functionality. To enable callcounters, you use the new
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; "callcounter" setting (for extension states in queue and subscriptions)
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; You are encouraged to use the dialplan groupcount functionality
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; to enforce call limits instead of using this channel-specific method.
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; You can still set limits per device in sip.conf or in a database by using
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; "setvar" to set variables that can be used in the dialplan for various limits.
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[general]
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context=public ; Default context for incoming calls. Defaults to 'default'
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;allowguest=no ; Allow or reject guest calls (default is yes)
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; If your Asterisk is connected to the Internet
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; and you have allowguest=yes
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; you want to check which services you offer everyone
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; out there, by enabling them in the default context (see below).
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;match_auth_username=yes ; if available, match user entry using the
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; 'username' field from the authentication line
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; instead of the From: field.
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allowoverlap=no ; Disable overlap dialing support. (Default is yes)
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;allowoverlap=yes ; Enable RFC3578 overlap dialing support.
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; Can use the Incomplete application to collect the
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; needed digits from an ambiguous dialplan match.
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;allowoverlap=dtmf ; Enable overlap dialing support using DTMF delivery
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; methods (inband, RFC2833, SIP INFO) in the early
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; media phase. Uses the Incomplete application to
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; collect the needed digits.
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;allowtransfer=no ; Disable all transfers (unless enabled in peers or users)
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; Default is enabled. The Dial() options 't' and 'T' are not
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; related as to whether SIP transfers are allowed or not.
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;realm=mydomain.tld ; Realm for digest authentication
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; defaults to "asterisk". If you set a system name in
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; asterisk.conf, it defaults to that system name
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; Realms MUST be globally unique according to RFC 3261
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; Set this to your host name or domain name
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;domainsasrealm=no ; Use domains list as realms
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; You can serve multiple Realms specifying several
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; 'domain=...' directives (see below).
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; In this case Realm will be based on request 'From'/'To' header
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; and should match one of domain names.
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; Otherwise default 'realm=...' will be used.
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;recordonfeature=automixmon ; Default feature to use when receiving 'Record: on' header
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; from an INFO message. Defaults to 'automon'. Works with
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; dynamic features. Feature must be usable on requesting
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; channel for it to work. Setting this value to a blank
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; will disable it.
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;recordofffeature=automixmon ; Default feature to use when receiving 'Record: off' header
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; from an INFO message. Defaults to 'automon'. Works with
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; dynamic features. Feature must be usable on requesting
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; channel for it to work. Setting this value to a blank
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; will disable it.
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; With the current situation, you can do one of four things:
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; a) Listen on a specific IPv4 address. Example: bindaddr=192.0.2.1
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; b) Listen on a specific IPv6 address. Example: bindaddr=2001:db8::1
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; c) Listen on the IPv4 wildcard. Example: bindaddr=0.0.0.0
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; d) Listen on the IPv4 and IPv6 wildcards. Example: bindaddr=::
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; (You can choose independently for UDP, TCP, and TLS, by specifying different values for
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; "udpbindaddr", "tcpbindaddr", and "tlsbindaddr".)
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; (Note that using bindaddr=:: will show only a single IPv6 socket in netstat.
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; IPv4 is supported at the same time using IPv4-mapped IPv6 addresses.)
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;
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; You may optionally add a port number. (The default is port 5060 for UDP and TCP, 5061
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; for TLS).
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; IPv4 example: bindaddr=0.0.0.0:5062
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; IPv6 example: bindaddr=[::]:5062
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;
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; The address family of the bound UDP address is used to determine how Asterisk performs
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; DNS lookups. In cases a) and c) above, only A records are considered. In case b), only
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; AAAA records are considered. In case d), both A and AAAA records are considered. Note,
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; however, that Asterisk ignores all records except the first one. In case d), when both A
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; and AAAA records are available, either an A or AAAA record will be first, and which one
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; depends on the operating system. On systems using glibc, AAAA records are given
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; priority.
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udpbindaddr=0.0.0.0 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all)
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; Optionally add a port number, 192.168.1.1:5062 (default is port 5060)
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; When a dialog is started with another SIP endpoint, the other endpoint
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; should include an Allow header telling us what SIP methods the endpoint
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; implements. However, some endpoints either do not include an Allow header
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; or lie about what methods they implement. In the former case, Asterisk
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; makes the assumption that the endpoint supports all known SIP methods.
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; If you know that your SIP endpoint does not provide support for a specific
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; method, then you may provide a comma-separated list of methods that your
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; endpoint does not implement in the disallowed_methods option. Note that
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; if your endpoint is truthful with its Allow header, then there is no need
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; to set this option. This option may be set in the general section or may
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; be set per endpoint. If this option is set both in the general section and
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; in a peer section, then the peer setting completely overrides the general
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; setting (i.e. the result is *not* the union of the two options).
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;
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; Note also that while Asterisk currently will parse an Allow header to learn
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; what methods an endpoint supports, the only actual use for this currently
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; is for determining if Asterisk may send connected line UPDATE requests and
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; MESSAGE requests. Its use may be expanded in the future.
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;
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; disallowed_methods = UPDATE
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;
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; Note that the TCP and TLS support for chan_sip is currently considered
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; experimental. Since it is new, all of the related configuration options are
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; subject to change in any release. If they are changed, the changes will
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; be reflected in this sample configuration file, as well as in the UPGRADE.txt file.
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;
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tcpenable=no ; Enable server for incoming TCP connections (default is no)
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tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces)
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; Optionally add a port number, 192.168.1.1:5062 (default is port 5060)
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;tlsenable=no ; Enable server for incoming TLS (secure) connections (default is no)
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;tlsbindaddr=0.0.0.0 ; IP address for TLS server to bind to (0.0.0.0) binds to all interfaces)
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; Optionally add a port number, 192.168.1.1:5063 (default is port 5061)
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; Remember that the IP address must match the common name (hostname) in the
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; certificate, so you don't want to bind a TLS socket to multiple IP addresses.
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; For details how to construct a certificate for SIP see
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; http://tools.ietf.org/html/draft-ietf-sip-domain-certs
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;tcpauthtimeout = 30 ; tcpauthtimeout specifies the maximum number
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; of seconds a client has to authenticate. If
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; the client does not authenticate beofre this
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; timeout expires, the client will be
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; disconnected. (default: 30 seconds)
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;tcpauthlimit = 100 ; tcpauthlimit specifies the maximum number of
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; unauthenticated sessions that will be allowed
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; to connect at any given time. (default: 100)
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;websocket_write_timeout = 100 ; Default write timeout to set on websocket transports.
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; This value may need to be adjusted for connections where
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; Asterisk must write a substantial amount of data and the
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; receiving clients are slow to process the received information.
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; Value is in milliseconds; default is 100 ms.
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transport=udp ; Set the default transports. The order determines the primary default transport.
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; If tcpenable=no and the transport set is tcp, we will fallback to UDP.
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srvlookup=yes ; Enable DNS SRV lookups on outbound calls
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; Note: Asterisk only uses the first host
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; in SRV records
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; Disabling DNS SRV lookups disables the
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; ability to place SIP calls based on domain
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; names to some other SIP users on the Internet
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; Specifying a port in a SIP peer definition or
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; when dialing outbound calls will supress SRV
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; lookups for that peer or call.
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;pedantic=yes ; Enable checking of tags in headers,
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; international character conversions in URIs
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; and multiline formatted headers for strict
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; SIP compatibility (defaults to "yes")
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; See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for a description of these parameters.
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;tos_sip=cs3 ; Sets TOS for SIP packets.
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;tos_audio=ef ; Sets TOS for RTP audio packets.
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;tos_video=af41 ; Sets TOS for RTP video packets.
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;tos_text=af41 ; Sets TOS for RTP text packets.
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;cos_sip=3 ; Sets 802.1p priority for SIP packets.
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;cos_audio=5 ; Sets 802.1p priority for RTP audio packets.
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;cos_video=4 ; Sets 802.1p priority for RTP video packets.
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;cos_text=3 ; Sets 802.1p priority for RTP text packets.
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;maxexpiry=3600 ; Maximum allowed time of incoming registrations (seconds)
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;minexpiry=60 ; Minimum length of registrations (default 60)
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;defaultexpiry=120 ; Default length of incoming/outgoing registration
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;submaxexpiry=3600 ; Maximum allowed time of incoming subscriptions (seconds), default: maxexpiry
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;subminexpiry=60 ; Minimum length of subscriptions, default: minexpiry
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;mwiexpiry=3600 ; Expiry time for outgoing MWI subscriptions
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;maxforwards=70 ; Setting for the SIP Max-Forwards: header (loop prevention)
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; Default value is 70
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;qualifyfreq=60 ; Qualification: How often to check for the host to be up in seconds
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; and reported in milliseconds with sip show settings.
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; Set to low value if you use low timeout for NAT of UDP sessions
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; Default: 60
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;qualifygap=100 ; Number of milliseconds between each group of peers being qualified
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; Default: 100
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;qualifypeers=1 ; Number of peers in a group to be qualified at the same time
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; Default: 1
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;keepalive=60 ; Interval at which keepalive packets should be sent to a peer
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; Valid options are yes (60 seconds), no, or the number of seconds.
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; Default: 0
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;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY
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;buggymwi=no ; Cisco SIP firmware doesn't support the MWI RFC
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; fully. Enable this option to not get error messages
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; when sending MWI to phones with this bug.
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;mwi_from=asterisk ; When sending MWI NOTIFY requests, use this setting in
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; the From: header as the "name" portion. Also fill the
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; "user" portion of the URI in the From: header with this
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; value if no fromuser is set
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; Default: empty
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;vmexten=voicemail ; dialplan extension to reach mailbox sets the
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; Message-Account in the MWI notify message
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; defaults to "asterisk"
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; Codec negotiation
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;
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; When Asterisk is receiving a call, the codec will initially be set to the
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; first codec in the allowed codecs defined for the user receiving the call
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; that the caller also indicates that it supports. But, after the caller
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; starts sending RTP, Asterisk will switch to using whatever codec the caller
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; is sending.
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;
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; When Asterisk is placing a call, the codec used will be the first codec in
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; the allowed codecs that the callee indicates that it supports. Asterisk will
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; *not* switch to whatever codec the callee is sending.
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;
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;preferred_codec_only=yes ; Respond to a SIP invite with the single most preferred codec
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; rather than advertising all joint codec capabilities. This
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; limits the other side's codec choice to exactly what we prefer.
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;disallow=all ; First disallow all codecs
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;allow=ulaw ; Allow codecs in order of preference
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;allow=ilbc ; see https://wiki.asterisk.org/wiki/display/AST/RTP+Packetization
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; for framing options
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;autoframing=yes ; Set packetization based on the remote endpoint's (ptime)
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; preferences. Defaults to no.
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;
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; This option specifies a preference for which music on hold class this channel
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; should listen to when put on hold if the music class has not been set on the
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; channel with Set(CHANNEL(musicclass)=whatever) in the dialplan, and the peer
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; channel putting this one on hold did not suggest a music class.
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;
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; This option may be specified globally, or on a per-user or per-peer basis.
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;
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;mohinterpret=default
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;
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; This option specifies which music on hold class to suggest to the peer channel
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; when this channel places the peer on hold. It may be specified globally or on
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; a per-user or per-peer basis.
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;
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;mohsuggest=default
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;
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;parkinglot=plaza ; Sets the default parking lot for call parking
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; This may also be set for individual users/peers
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; Parkinglots are configured in features.conf
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;language=en ; Default language setting for all users/peers
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; This may also be set for individual users/peers
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;tonezone=se ; Default tonezone for all users/peers
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; This may also be set for individual users/peers
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;relaxdtmf=yes ; Relax dtmf handling
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;trustrpid = no ; If Remote-Party-ID should be trusted
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;sendrpid = yes ; If Remote-Party-ID should be sent (defaults to no)
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;sendrpid = rpid ; Use the "Remote-Party-ID" header
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; to send the identity of the remote party
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; This is identical to sendrpid=yes
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;sendrpid = pai ; Use the "P-Asserted-Identity" header
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; to send the identity of the remote party
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;rpid_update = no ; In certain cases, the only method by which a connected line
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; change may be immediately transmitted is with a SIP UPDATE request.
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; If communicating with another Asterisk server, and you wish to be able
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; transmit such UPDATE messages to it, then you must enable this option.
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; Otherwise, we will have to wait until we can send a reinvite to
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; transmit the information.
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;trust_id_outbound = no ; Controls whether or not we trust this peer with private identity
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; information (when the remote party has callingpres=prohib or equivalent).
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; no - RPID/PAI headers will not be included for private peer information
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; yes - RPID/PAI headers will include the private peer information. Privacy
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; requirements will be indicated in a Privacy header for sendrpid=pai
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; legacy - RPID/PAI will be included for private peer information. In the
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; case of sendrpid=pai, private data that would be included in them
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; will be anonymized. For sendrpid=rpid, private data may be included
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; but the remote party's domain will be anonymized. The way legacy
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; behaves may violate RFC-3325, but it follows historic behavior.
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; This option is set to 'legacy' by default
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;prematuremedia=no ; Some ISDN links send empty media frames before
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; the call is in ringing or progress state. The SIP
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; channel will then send 183 indicating early media
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; which will be empty - thus users get no ring signal.
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; Setting this to "yes" will stop any media before we have
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; call progress (meaning the SIP channel will not send 183 Session
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; Progress for early media). Default is "yes". Also make sure that
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; the SIP peer is configured with progressinband=never.
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;
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; In order for "noanswer" applications to work, you need to run
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; the progress() application in the priority before the app.
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;progressinband=never ; If we should generate in-band ringing always
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; use 'never' to never use in-band signalling, even in cases
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; where some buggy devices might not render it
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; Valid values: yes, no, never Default: never
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;useragent=Asterisk PBX ; Allows you to change the user agent string
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; The default user agent string also contains the Asterisk
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; version. If you don't want to expose this, change the
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; useragent string.
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;promiscredir = no ; If yes, allows 302 or REDIR to non-local SIP address
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; Note that promiscredir when redirects are made to the
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; local system will cause loops since Asterisk is incapable
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; of performing a "hairpin" call.
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;usereqphone = no ; If yes, ";user=phone" is added to uri that contains
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; a valid phone number
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;dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. Default: rfc2833
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; Other options:
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; info : SIP INFO messages (application/dtmf-relay)
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; shortinfo : SIP INFO messages (application/dtmf)
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; inband : Inband audio (requires 64 kbit codec -alaw, ulaw)
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; auto : Use rfc2833 if offered, inband otherwise
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;compactheaders = yes ; send compact sip headers.
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;
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;videosupport=yes ; Turn on support for SIP video. You need to turn this
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; on in this section to get any video support at all.
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|
; 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]
|
|
|
|
;textsupport=no ; Support for ITU-T T.140 realtime text.
|
|
; The default value is "no".
|
|
|
|
;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.
|
|
; This option is set to "yes" by default.
|
|
|
|
;auth_options_requests = yes ; Enabling this option will authenticate OPTIONS requests just like
|
|
; INVITE requests are. By default this option is disabled.
|
|
|
|
;accept_outofcall_message = no ; Disable this option to reject all MESSAGE requests outside of a
|
|
; call. By default, this option is enabled. When enabled, MESSAGE
|
|
; requests are passed in to the dialplan.
|
|
|
|
;outofcall_message_context = messages ; Context all out of dialog msgs are sent to. When this
|
|
; option is not set, the context used during peer matching
|
|
; is used. This option can be defined at both the peer and
|
|
; global level.
|
|
|
|
;auth_message_requests = yes ; Enabling this option will authenticate MESSAGE requests.
|
|
; By default this option is enabled. However, it can be disabled
|
|
; should an application desire to not load the Asterisk server with
|
|
; doing authentication and implement end to end security in the
|
|
; message body.
|
|
|
|
;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
|
|
;outboundproxy=192.0.2.1 ; IPv4 address literal (default port is 5060)
|
|
;outboundproxy=2001:db8::1 ; IPv6 address literal (default port is 5060)
|
|
;outboundproxy=192.168.0.2.1:5062 ; IPv4 address literal with explicit port
|
|
;outboundproxy=[2001:db8::1]:5062 ; IPv6 address literal with explicit port
|
|
; ; (could also be tcp,udp) - defining transports on the proxy line only
|
|
; ; applies for the global proxy, otherwise use the transport= option
|
|
;matchexternaddrlocally = yes ; Only substitute the externaddr 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.
|
|
;contactacl=named_acl_example ; Use named ACLs defined in acl.conf
|
|
|
|
;rtp_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
|
|
;legacy_useroption_parsing=yes ; Default "no" ; If you have this option enabled and there are semicolons
|
|
; in the user field of a sip URI, the field be truncated
|
|
; at the first semicolon seen. This effectively makes
|
|
; semicolon a non-usable character for peer names, extensions,
|
|
; and maybe other, less tested things. This can be useful
|
|
; for improving compatability with devices that like to use
|
|
; user options for whatever reason. The behavior is similar to
|
|
; how SIP URI's were typically handled in 1.6.2, hence the name.
|
|
|
|
;send_diversion=no ; Default "yes" ; Asterisk normally sends Diversion headers with certain SIP
|
|
; invites to relay data about forwarded calls. If this option
|
|
; is disabled, Asterisk won't send Diversion headers unless
|
|
; they are added manually.
|
|
|
|
; The shrinkcallerid function removes '(', ' ', ')', non-trailing '.', and '-' not
|
|
; in square brackets. For example, the caller id value 555.5555 becomes 5555555
|
|
; when this option is enabled. Disabling this option results in no modification
|
|
; of the caller id value, which is necessary when the caller id represents something
|
|
; that must be preserved. This option can only be used in the [general] section.
|
|
; By default this option is on.
|
|
;
|
|
;shrinkcallerid=yes ; on by default
|
|
|
|
|
|
;use_q850_reason = no ; Default "no"
|
|
; Set to yes add Reason header and use Reason header if it is available.
|
|
|
|
; When the Transfer() application sends a REFER SIP message, extra headers specified in
|
|
; the dialplan by way of SIPAddHeader are sent out with that message. 1.8 and earlier did not
|
|
; add the extra headers. To revert to 1.8- behavior, call SIPRemoveHeader with no arguments
|
|
; before calling Transfer() to remove all additional headers from the channel. The setting
|
|
; below is for transitional compatibility only.
|
|
;
|
|
;refer_addheaders=yes ; on by default
|
|
|
|
;autocreatepeer=no ; Allow any UAC not explicitly defined to register
|
|
; WITHOUT AUTHENTICATION. Enabling this options poses a high
|
|
; potential security risk and should be avoided unless the
|
|
; server is behind a trusted firewall.
|
|
; If set to "yes", then peers created in this fashion
|
|
; are purged during SIP reloads.
|
|
; When set to "persist", the peers created in this fashion
|
|
; are not purged during SIP reloads.
|
|
|
|
;
|
|
;------------------------ TLS settings ------------------------------------------------------------
|
|
;tlscertfile=</path/to/certificate.pem> ; Certificate chain (*.pem format only) to use for TLS connections
|
|
; The certificates must be sorted starting with the subject's certificate
|
|
; and followed by intermediate CA certificates if applicable.
|
|
; Default is to look for "asterisk.pem" in current directory
|
|
|
|
;tlsprivatekey=</path/to/private.pem> ; Private key file (*.pem format 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.
|
|
|
|
;tlscapath=</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.
|
|
;
|
|
;--------------------------- 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'.
|
|
; uac - Default to the caller initially refreshing when possible
|
|
; uas - Default to the callee initially refreshing when possible
|
|
;
|
|
; Note that, due to recommendations in RFC 4028, Asterisk will always honor the other
|
|
; endpoint's preference for who will handle refreshes. Asterisk will never override the
|
|
; preferences of the other endpoint. Doing so could result in Asterisk and the endpoint
|
|
; fighting over who sends the refreshes. This holds true for the initiation of session
|
|
; timers and subsequent re-INVITE requests whether Asterisk is the caller or callee, or
|
|
; whether Asterisk is currently the refresher or not.
|
|
;
|
|
;session-timers=originate
|
|
;session-expires=600
|
|
;session-minse=90
|
|
;session-refresher=uac
|
|
;
|
|
;--------------------------- SIP DEBUGGING ---------------------------------------------------
|
|
;sipdebug = yes ; Turn on SIP debugging by default, from
|
|
; the moment the channel loads this configuration.
|
|
; NOTE: You cannot use the CLI to turn it off. You'll
|
|
; need to edit this and reload the config.
|
|
;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 SUPPORT ----------------------------------
|
|
;
|
|
; This setting is available in the [general] section as well as in device configurations.
|
|
; Setting this to yes enables T.38 FAX (UDPTL) on 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.
|
|
;
|
|
; In some cases, T.38 endpoints will provide a T38FaxMaxDatagram value (during T.38 setup) that
|
|
; is based on an incorrect interpretation of the T.38 recommendation, and results in failures
|
|
; because Asterisk does not believe it can send T.38 packets of a reasonable size to that
|
|
; endpoint (Cisco media gateways are one example of this situation). In these cases, during a
|
|
; T.38 call you will see warning messages on the console/in the logs from the Asterisk UDPTL
|
|
; stack complaining about lack of buffer space to send T.38 FAX packets. If this occurs, you
|
|
; can set an override (globally, or on a per-device basis) to make Asterisk ignore the
|
|
; T38FaxMaxDatagram value specified by the other endpoint, and use a configured value instead.
|
|
; This can be done by appending 'maxdatagram=<value>' to the t38pt_udptl configuration option,
|
|
; like this:
|
|
;
|
|
; t38pt_udptl = yes,fec,maxdatagram=400 ; Enables T.38 with FEC error correction and overrides
|
|
; ; the other endpoint's provided value to assume we can
|
|
; ; send 400 byte T.38 FAX packets to it.
|
|
;
|
|
; FAX detection will cause the SIP channel to jump to the 'fax' extension (if it exists)
|
|
; based one or more events being detected. The events that can be detected are an incoming
|
|
; CNG tone or an incoming T.38 re-INVITE request.
|
|
;
|
|
; faxdetect = yes ; Default 'no', 'yes' enables both CNG and T.38 detection
|
|
; faxdetect = cng ; Enables only CNG detection
|
|
; faxdetect = t38 ; Enables only T.38 detection
|
|
;
|
|
;----------------------------------------- OUTBOUND SIP REGISTRATIONS ------------------------
|
|
; Asterisk can register as a SIP user agent to a SIP proxy (provider)
|
|
; Format for the register statement is:
|
|
; register => [peer?][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
|
|
;
|
|
;register => tls://username:xxxxxx@sip-tls-proxy.example.org
|
|
;
|
|
; The 'transport' part defaults to 'udp' but may also be 'tcp', 'tls', 'ws', or 'wss'.
|
|
; Using 'udp://' explicitly is also useful in case the username part
|
|
; contains a '/' ('user/name').
|
|
|
|
;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
|
|
;register_retry_403=yes ; Treat 403 responses to registrations as if they were
|
|
; 401 responses and continue retrying according to normal
|
|
; retry rules.
|
|
|
|
;----------------------------------------- OUTBOUND MWI SUBSCRIPTIONS -------------------------
|
|
; Asterisk can subscribe to receive the MWI from another SIP server and store it locally for retrieval
|
|
; by other phones. At this time, you can only subscribe using UDP as the transport.
|
|
; Format for the mwi register statement is:
|
|
; mwi => user[:secret[:authuser]]@host[:port]/mailbox
|
|
;
|
|
; Examples:
|
|
;mwi => 1234:password@mysipprovider.com/1234
|
|
;mwi => 1234:password@myportprovider.com:6969/1234
|
|
;mwi => 1234:password:authuser@myauthprovider.com/1234
|
|
;mwi => 1234:password:authuser@myauthportprovider.com:6969/1234
|
|
;
|
|
; 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. "externaddr = 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 port specified in the "udpbindaddr"
|
|
; (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:
|
|
;
|
|
; externaddr = 12.34.56.78 ; use this address.
|
|
; externaddr = 12.34.56.78:9900 ; use this address and port.
|
|
; externaddr = mynat.my.org:12600 ; Public address of my nat box.
|
|
; externtcpport = 9900 ; The externally mapped tcp port, when Asterisk is behind a static NAT or PAT.
|
|
; ; externtcpport will default to the externaddr or externhost port if either one is set.
|
|
; externtlsport = 12600 ; The externally mapped tls port, when Asterisk is behind a static NAT or PAT.
|
|
; ; externtlsport port will default to the RFC designated port of 5061.
|
|
;
|
|
; b. "externhost = hostname[:port]" is similar to "externaddr" 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
|
|
;
|
|
; Note that at the moment all these mechanism work only for the SIP socket.
|
|
; The IP address discovered with externaddr/externhost 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 "externaddr" and
|
|
; "externhost" might not help you configure addresses properly.
|
|
;
|
|
; NOTE 2: when using "externaddr" or "externhost", the address part is
|
|
; also used as the external address for media sessions. Thus, the port
|
|
; information in the SDP may be wrong!
|
|
;
|
|
; 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 ; Do no special NAT handling other than RFC3581
|
|
; nat = force_rport ; Pretend there was an rport parameter even if there wasn't
|
|
; nat = comedia ; Send media to the port Asterisk received it from regardless
|
|
; ; of where the SDP says to send it.
|
|
; nat = auto_force_rport ; Set the force_rport option if Asterisk detects NAT (default)
|
|
; nat = auto_comedia ; Set the comedia option if Asterisk detects NAT
|
|
;
|
|
; The nat settings can be combined. For example, to set both force_rport and comedia
|
|
; one would set nat=force_rport,comedia. If any of the comma-separated options is 'no',
|
|
; Asterisk will ignore any other settings and set nat=no. If one of the "auto" settings
|
|
; is used in conjunction with its non-auto counterpart (nat=comedia,auto_comedia), then
|
|
; the non-auto option will be ignored.
|
|
;
|
|
; The RFC 3581-defined 'rport' parameter allows a client to request that Asterisk send
|
|
; SIP responses to it via the source IP and port from which the request originated
|
|
; instead of the address/port listed in the top-most Via header. This is useful if a
|
|
; client knows that it is behind a NAT and therefore cannot guess from what address/port
|
|
; its request will be sent. Asterisk will always honor the 'rport' parameter if it is
|
|
; sent. The force_rport setting causes Asterisk to always send responses back to the
|
|
; address/port from which it received requests; even if the other side doesn't support
|
|
; adding the 'rport' parameter.
|
|
;
|
|
; 'comedia RTP handling' refers to the technique of sending RTP to the port that the
|
|
; the other endpoint's RTP arrived from, and means 'connection-oriented media'. This is
|
|
; only partially related to RFC 4145 which was referred to as COMEDIA while it was in
|
|
; draft form. This method is used to accomodate endpoints that may be located behind
|
|
; NAT devices, and as such the address/port they tell Asterisk to send RTP packets to
|
|
; for their media streams is not the actual address/port that will be used on the nearer
|
|
; side of the NAT.
|
|
;
|
|
; IT IS IMPORTANT TO NOTE that if the nat setting in the general section differs from
|
|
; the nat setting in a peer definition, then the peer username will be discoverable
|
|
; by outside parties as Asterisk will respond to different ports for defined and
|
|
; undefined peers. For this reason it is recommended to ONLY DEFINE NAT SETTINGS IN THE
|
|
; GENERAL SECTION. Specifically, if nat=force_rport in one section and nat=no in the
|
|
; other, then valid peers with settings differing from those in the general section will
|
|
; be discoverable.
|
|
;
|
|
; In addition to these settings, Asterisk *always* uses 'symmetric RTP' mode as defined by
|
|
; RFC 4961; Asterisk will always send RTP packets from the same port number it expects
|
|
; to receive them on.
|
|
;
|
|
; The IP address used for media (audio, video, and text) in the SDP can also be overridden by using
|
|
; the media_address configuration option. This is only applicable to the general section and
|
|
; can not be set per-user or per-peer.
|
|
;
|
|
; media_address = 172.16.42.1
|
|
;
|
|
; Through the use of the res_stun_monitor module, Asterisk has the ability to detect when the
|
|
; perceived external network address has changed. When the stun_monitor is installed and
|
|
; configured, chan_sip will renew all outbound registrations when the monitor detects any sort
|
|
; of network change has occurred. By default this option is enabled, but only takes effect once
|
|
; res_stun_monitor is configured. If res_stun_monitor is enabled and you wish to not
|
|
; generate all outbound registrations on a network change, use the option below to disable
|
|
; this feature.
|
|
;
|
|
; subscribe_network_change_event = yes ; on by default
|
|
;
|
|
; ICE/STUN/TURN usage can be enabled globally or on a per-peer basis using the icesupport
|
|
; configuration option. When set to yes ICE support is enabled. When set to no it is disabled.
|
|
; It is disabled by default.
|
|
;
|
|
; icesupport = yes
|
|
|
|
;----------------------------------- MEDIA HANDLING --------------------------------
|
|
; By default, Asterisk tries to re-invite media streams 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 well in the case where Asterisk is outside and the
|
|
; clients are on the inside of a NAT. In that case, you want to set directmedia=nonat.
|
|
;
|
|
;directmedia=yes ; Asterisk by default tries to redirect the
|
|
; RTP media stream 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 want 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).
|
|
|
|
; Additionally this option does not disable all reINVITE operations.
|
|
; It only controls Asterisk generating reINVITEs for the specific
|
|
; purpose of setting up a direct media path. If a reINVITE is
|
|
; needed to switch a media stream to inactive (when placed on
|
|
; hold) or to T.38, it will still be done, regardless of this
|
|
; setting. Note that direct T.38 is not supported.
|
|
|
|
;directmedia=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).
|
|
|
|
;directmedia=update ; Yet a third option... use UPDATE for media path redirection,
|
|
; instead of INVITE. This can be combined with 'nonat', as
|
|
; 'directmedia=update,nonat'. It implies 'yes'.
|
|
|
|
;directmedia=outgoing ; When sending directmedia reinvites, do not send an immediate
|
|
; reinvite on an incoming call leg. This option is useful when
|
|
; peered with another SIP user agent that is known to send
|
|
; immediate direct media reinvites upon call establishment. Setting
|
|
; the option in this situation helps to prevent potential glares.
|
|
; Setting this option implies 'yes'.
|
|
|
|
;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 directmedia is enabled when
|
|
; the device is actually behind NAT.
|
|
|
|
;directmediadeny=0.0.0.0/0 ; Use directmediapermit and directmediadeny to restrict
|
|
;directmediapermit=172.16.0.0/16; which peers should be able to pass directmedia to each other
|
|
; (There is no default setting, this is just an example)
|
|
; Use this if some of your phones are on IP addresses that
|
|
; can not reach each other directly. This way you can force
|
|
; RTP to always flow through asterisk in such cases.
|
|
;directmediaacl=acl_example ; Use named ACLs defined in acl.conf
|
|
|
|
;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.
|
|
|
|
;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
|
|
;encryption=no ; Whether to offer SRTP encrypted media (and only SRTP encrypted media)
|
|
; on outgoing calls to a peer. Calls will fail with HANGUPCAUSE=58 if
|
|
; the peer does not support SRTP. Defaults to no.
|
|
;encryption_taglen=80 ; Set the auth tag length offered in the INVITE either 32/80 default 80
|
|
;
|
|
;avpf=yes ; Enable inter-operability with media streams using the AVPF RTP profile.
|
|
; This will cause all offers and answers to use AVPF (or SAVPF). This
|
|
; option may be specified at the global or peer scope.
|
|
;force_avp=yes ; Force 'RTP/AVP', 'RTP/AVPF', 'RTP/SAVP', and 'RTP/SAVPF' to be used for
|
|
; media streams when appropriate, even if a DTLS stream is present.
|
|
;----------------------------------------- REALTIME SUPPORT ------------------------
|
|
; For additional information on ARA, the Asterisk Realtime Architecture,
|
|
; please read https://wiki.asterisk.org/wiki/display/AST/Realtime+Database+Configuration
|
|
;
|
|
;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.
|
|
|
|
;------------------------------ Advice of Charge CONFIGURATION --------------------------
|
|
; snom_aoc_enabled = yes; ; This options turns on and off support for sending AOC-D and
|
|
; AOC-E to snom endpoints. This option can be used both in the
|
|
; peer and global scope. The default for this option is off.
|
|
|
|
|
|
;------------------------------ 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.
|
|
|
|
; jbtargetextra = 40 ; This option only affects the jb when 'jbimpl = adaptive' is set.
|
|
; The option represents the number of milliseconds by which the new jitter buffer
|
|
; will pad its size. the default is 40, so without modification, the new
|
|
; jitter buffer will set its size to the jitter value plus 40 milliseconds.
|
|
; increasing this value may help if your network normally has low jitter,
|
|
; but occasionally has spikes.
|
|
|
|
; 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
|
|
;
|
|
; SIP entities have a 'type' which determines their roles within Asterisk.
|
|
; * For entities with 'type=peer':
|
|
; Peers handle both inbound and outbound calls and are matched by ip/port, so for
|
|
; The case of incoming calls from the peer, the IP address must match in order for
|
|
; The invitation to work. This means calls made from either direction won't work if
|
|
; The peer is unregistered while host=dynamic or if the host is otherise not set to
|
|
; the correct IP of the sender.
|
|
; * For entities with 'type=user':
|
|
; Asterisk users handle inbound calls only (meaning they call Asterisk, Asterisk can't
|
|
; call them) and are matched by their authorization information (authname and secret).
|
|
; Asterisk doesn't rely on their IP and will accept calls regardless of the host setting
|
|
; as long as the incoming SIP invite authorizes successfully.
|
|
; * For entities with 'type=friend':
|
|
; Asterisk will create the entity as both a friend and a peer. Asterisk will accept
|
|
; calls from friends like it would for users, requiring only that the authorization
|
|
; matches rather than the IP address. Since it is also a peer, a friend entity can
|
|
; be called as long as its IP is known to Asterisk. In the case of host=dynamic,
|
|
; this means it is necessary for the entity to register before Asterisk can call it.
|
|
;
|
|
; Use remotesecret for outbound authentication, and secret for authenticating
|
|
; inbound requests. For historical reasons, if no remotesecret is supplied for an
|
|
; outbound registration or call, the secret will be used.
|
|
;
|
|
; 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
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; --------------------
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; context
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; callingpres
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; permit
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; deny
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; secret
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; md5secret
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; remotesecret
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; transport
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; dtmfmode
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; directmedia
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; nat
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; callgroup
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; pickupgroup
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; language
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; allow
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; disallow
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; autoframing
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; insecure
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; trustrpid
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; trust_id_outbound
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; progressinband
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; promiscredir
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; useclientcode
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; accountcode
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; setvar
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; callerid
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; amaflags
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; callcounter
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; busylevel
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; allowoverlap
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; allowsubscribe
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; allowtransfer
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; ignoresdpversion
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; subscribecontext
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; template
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; videosupport
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; maxcallbitrate
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; rfc2833compensate
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; mailbox
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; session-timers
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; session-expires
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; session-minse
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; session-refresher
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; t38pt_usertpsource
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; regexten
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; fromdomain
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; fromuser
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; host
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; port
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; qualify
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; keepalive
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; defaultip
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; defaultuser
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; rtptimeout
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; rtpholdtimeout
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; sendrpid
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; outboundproxy
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; rfc2833compensate
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; callbackextension
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; timert1
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; timerb
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; qualifyfreq
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; t38pt_usertpsource
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; contactpermit ; Limit what a host may register as (a neat trick
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; contactdeny ; is to register at the same IP as a SIP provider,
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; contactacl ; then call oneself, and get redirected to that
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; ; same location).
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; directmediapermit
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; directmediadeny
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; directmediaacl
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; unsolicited_mailbox
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; use_q850_reason
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; maxforwards
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; encryption
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; description ; Used to provide a description of the peer in console output
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; dtlsenable
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; dtlsverify
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; dtlsrekey
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; dtlscertfile
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; dtlsprivatekey
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; dtlscipher
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; dtlscafile
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; dtlscapath
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; dtlssetup
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; dtlsfingerprint
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;
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|
|
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;------------------------------------------------------------------------------
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; DTLS-SRTP CONFIGURATION
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;
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; DTLS-SRTP support is available if the underlying RTP engine in use supports it.
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;
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; dtlsenable = yes ; Enable or disable DTLS-SRTP support
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; dtlsverify = yes ; Verify that provided peer certificate and fingerprint are valid
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; ; A value of 'yes' will perform both certificate and fingerprint verification
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; ; A value of 'no' will perform no certificate or fingerprint verification
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; ; A value of 'fingerprint' will perform ONLY fingerprint verification
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; ; A value of 'certificate' will perform ONLY certficiate verification
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; dtlsrekey = 60 ; Interval at which to renegotiate the TLS session and rekey the SRTP session
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; ; If this is not set or the value provided is 0 rekeying will be disabled
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; dtlscertfile = file ; Path to certificate file to present
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; dtlsprivatekey = file ; Path to private key for certificate file
|
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; dtlscipher = <SSL cipher string> ; Cipher to use for TLS negotiation
|
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; ; A list of valid SSL cipher strings can be found at:
|
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; ; http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS
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; dtlscafile = file ; Path to certificate authority certificate
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; dtlscapath = path ; Path to a directory containing certificate authority certificates
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; dtlssetup = actpass ; Whether we are willing to accept connections, connect to the other party, or both.
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; ; Valid options are active (we want to connect to the other party), passive (we want to
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; ; accept connections only), and actpass (we will do both). This value will be used in
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; ; the outgoing SDP when offering and for incoming SDP offers when the remote party sends
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; ; actpass
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; dtlsfingerprint = sha-1 ; The hash to use for the fingerprint in SDP (valid options are sha-1 and sha-256)
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|
|
;[sip_proxy]
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; For incoming calls only. Example: FWD (Free World Dialup)
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; We match on IP address of the proxy for incoming calls
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|
; since we can not match on username (caller id)
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|
;type=peer
|
|
;context=from-fwd
|
|
;host=fwd.pulver.com
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|
|
|
;[sip_proxy-out]
|
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;type=peer ; we only want to call out, not be called
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|
;remotesecret=guessit ; Our password to their service
|
|
;defaultuser=yourusername ; Authentication user for outbound proxies
|
|
;fromuser=yourusername ; Many SIP providers require this!
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|
;fromdomain=provider.sip.domain
|
|
;host=box.provider.com
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|
;transport=udp,tcp ; This sets the default transport type to udp for outgoing, and will
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|
; ; accept both tcp and udp. The default transport type is only used for
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|
; ; outbound messages until a Registration takes place. During the
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|
; ; peer Registration the transport type may change to another supported
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; ; type if the peer requests so.
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|
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;usereqphone=yes ; This provider requires ";user=phone" on URI
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|
;callcounter=yes ; Enable call counter
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|
;busylevel=2 ; Signal busy at 2 or more calls
|
|
;outboundproxy=proxy.provider.domain ; send outbound signaling to this proxy, not directly to the peer
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|
;port=80 ; The port number we want to connect to on the remote side
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; Also used as "defaultport" in combination with "defaultip" settings
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|
|
|
;--- sample definition for a provider
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|
;[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
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|
; ; accept both tcp and udp. Default is udp. The first transport
|
|
; ; listed will always be used for outgoing connections.
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|
;unsolicited_mailbox=4015552299 ; If the remote SIP server sends an unsolicited MWI NOTIFY message the new/old
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; ; message count will be stored in the configured virtual mailbox. It can be used
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; ; by any device supporting MWI by specifying <configured value>@SIP_Remote as the
|
|
; ; mailbox.
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|
|
|
;
|
|
; Because you might have a large number of similar sections, it is generally
|
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; 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:
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|
[basic-options](!) ; a template
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dtmfmode=rfc2833
|
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context=from-office
|
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type=friend
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[natted-phone](!,basic-options) ; another template inheriting basic-options
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directmedia=no
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host=dynamic
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[public-phone](!,basic-options) ; another template inheriting basic-options
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directmedia=yes
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|
[my-codecs](!) ; a template for my preferred codecs
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|
disallow=all
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|
allow=ilbc
|
|
allow=g729
|
|
allow=gsm
|
|
allow=g723
|
|
allow=ulaw
|
|
; Or, more simply:
|
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;allow=!all,ilbc,g729,gsm,g723,ulaw
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|
|
[ulaw-phone](!) ; and another one for ulaw-only
|
|
disallow=all
|
|
allow=ulaw
|
|
; Again, more simply:
|
|
;allow=!all,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
|
|
;recordonfeature=dynamicfeature1 ; Feature to use when INFO with Record: on is received.
|
|
;recordofffeature=dynamicfeature2 ; Feature to use when INFO with Record: off is received.
|
|
;callerid=John Doe <1234> ; Full caller ID, to override the phones config
|
|
; on incoming calls to Asterisk
|
|
;description=Courtesy Phone ; Description of the peer. Shown when doing 'sip show peers'.
|
|
;host=192.168.0.23 ; we have a static but private IP address
|
|
; No registration allowed
|
|
;directmedia=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 function CALLERPRES documentation for possible
|
|
; values.
|
|
|
|
;[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
|
|
;directmedia=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
|
|
;namedcallgroup=engineering,sales,netgroup,protgroup ; We are in named call groups engineering,sales,netgroup,protgroup
|
|
;namedpickupgroup=sales ; We can do call pick-p for named call group sales
|
|
;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
|
|
;permit=192.168.0.60/24 ; we can also use CIDR notation for subnet masks
|
|
;permit=2001:db8::/32 ; IPv6 ACLs can be specified if desired. IPv6 ACLs
|
|
; apply only to IPv6 addresses, and IPv4 ACLs apply
|
|
; only to IPv4 addresses.
|
|
;acl=named_acl_example ; Use named ACLs defined in acl.conf
|
|
|
|
;[cisco1]
|
|
;type=friend
|
|
;secret=blah
|
|
;qualify=200 ; Qualify peer is no more than 200ms away
|
|
;host=dynamic ; This device registers with us
|
|
;directmedia=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.
|