;
; SIP Configuration example for Asterisk
;
+; Note: Please read the security documentation for Asterisk in order to
+; understand the risks of installing Asterisk with the sample
+; configuration. If your Asterisk is installed on a public
+; IP address connected to the Internet, you will want to learn
+; about the various security settings BEFORE you start
+; Asterisk.
+;
+; Especially note the following settings:
+; - allowguest (default enabled)
+; - permit/deny - IP address filters
+; - contactpermit/contactdeny - IP address filters for registrations
+; - context - Which set of services you offer various users
+;
; SIP dial strings
;-----------------------------------------------------------
; In the dialplan (extensions.conf) you can use several
; SIP/username@domain (SIP uri)
; SIP/username[:password[:md5secret[:authname[:transport]]]]@host[:port]
; SIP/devicename/extension
+; SIP/devicename/extension/IPorHost
+; SIP/username@domain//IPorHost
;
;
; Devicename
; SIP/sales:topsecret::account02@domain.com:5062
; SIP/12345678::bc53f0ba8ceb1ded2b70e05c3f91de4f:myname@192.168.0.1
;
+; IPorHost
+; The next server for this call regardless of domain/peer
+;
; All of these dial strings specify the SIP request URI.
; In addition, you can specify a specific To: header by adding an
; exclamation mark after the dial string, like
;
; SIP/sales@mysipproxy!sales@edvina.net
;
+; A new feature for 1.8 allows one to specify a host or IP address to use
+; when routing the call. This is typically used in tandem with func_srv if
+; multiple methods of reaching the same domain exist. The host or IP address
+; is specified after the third slash in the dialstring. Examples:
+;
+; SIP/devicename/extension/IPorHost
+; SIP/username@domain//IPorHost
+;
; CLI Commands
; -------------------------------------------------------------
; Useful CLI commands to check peers/users:
;
; sip set debug on Show all SIP messages
;
-; module reload chan_sip.so Reload configuration file
+; sip reload Reload configuration file
+; sip show settings Show the current channel configuration
;
;------- Naming devices ------------------------------------------------------
;
; combination with the "defaultip" setting.
;-----------------------------------------------------------------------------
-; ** Deprecated configuration options **
-; The "call-limit" configuation option is deprecated. It still works in
-; this version of Asterisk, but will disappear in the next version.
+; ** Old configuration options **
+; The "call-limit" configuation option is considered old is replaced
+; by new functionality. To enable callcounters, you use the new
+; "callcounter" setting (for extension states in queue and subscriptions)
; You are encouraged to use the dialplan groupcount functionality
; to enforce call limits instead of using this channel-specific method.
-;
; You can still set limits per device in sip.conf or in a database by using
; "setvar" to set variables that can be used in the dialplan for various limits.
[general]
-context=default ; Default context for incoming calls
+context=public ; Default context for incoming calls. Defaults to 'default'
;allowguest=no ; Allow or reject guest calls (default is yes)
+ ; If your Asterisk is connected to the Internet
+ ; and you have allowguest=yes
+ ; you want to check which services you offer everyone
+ ; out there, by enabling them in the default context (see below).
;match_auth_username=yes ; if available, match user entry using the
; 'username' field from the authentication line
; instead of the From: field.
allowoverlap=no ; Disable overlap dialing support. (Default is yes)
+;allowoverlap=yes ; Enable RFC3578 overlap dialing support.
+ ; Can use the Incomplete application to collect the
+ ; needed digits from an ambiguous dialplan match.
+;allowoverlap=dtmf ; Enable overlap dialing support using DTMF delivery
+ ; methods (inband, RFC2833, SIP INFO) in the early
+ ; media phase. Uses the Incomplete application to
+ ; collect the needed digits.
;allowtransfer=no ; Disable all transfers (unless enabled in peers or users)
- ; Default is enabled
+ ; Default is enabled. The Dial() options 't' and 'T' are not
+ ; related as to whether SIP transfers are allowed or not.
;realm=mydomain.tld ; Realm for digest authentication
; defaults to "asterisk". If you set a system name in
; asterisk.conf, it defaults to that system name
; Realms MUST be globally unique according to RFC 3261
; Set this to your host name or domain name
+;domainsasrealm=no ; Use domains list as realms
+ ; You can serve multiple Realms specifying several
+ ; 'domain=...' directives (see below).
+ ; In this case Realm will be based on request 'From'/'To' header
+ ; and should match one of domain names.
+ ; Otherwise default 'realm=...' will be used.
+
+; With the current situation, you can do one of four things:
+; a) Listen on a specific IPv4 address. Example: bindaddr=192.0.2.1
+; b) Listen on a specific IPv6 address. Example: bindaddr=2001:db8::1
+; c) Listen on the IPv4 wildcard. Example: bindaddr=0.0.0.0
+; d) Listen on the IPv4 and IPv6 wildcards. Example: bindaddr=::
+; (You can choose independently for UDP, TCP, and TLS, by specifying different values for
+; "udpbindaddr", "tcpbindaddr", and "tlsbindaddr".)
+; (Note that using bindaddr=:: will show only a single IPv6 socket in netstat.
+; IPv4 is supported at the same time using IPv4-mapped IPv6 addresses.)
+;
+; You may optionally add a port number. (The default is port 5060 for UDP and TCP, 5061
+; for TLS).
+; IPv4 example: bindaddr=0.0.0.0:5062
+; IPv6 example: bindaddr=[::]:5062
+;
+; The address family of the bound UDP address is used to determine how Asterisk performs
+; DNS lookups. In cases a) and c) above, only A records are considered. In case b), only
+; AAAA records are considered. In case d), both A and AAAA records are considered. Note,
+; however, that Asterisk ignores all records except the first one. In case d), when both A
+; and AAAA records are available, either an A or AAAA record will be first, and which one
+; depends on the operating system. On systems using glibc, AAAA records are given
+; priority.
+
udpbindaddr=0.0.0.0 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all)
; Optionally add a port number, 192.168.1.1:5062 (default is port 5060)
+; When a dialog is started with another SIP endpoint, the other endpoint
+; should include an Allow header telling us what SIP methods the endpoint
+; implements. However, some endpoints either do not include an Allow header
+; or lie about what methods they implement. In the former case, Asterisk
+; makes the assumption that the endpoint supports all known SIP methods.
+; If you know that your SIP endpoint does not provide support for a specific
+; method, then you may provide a comma-separated list of methods that your
+; endpoint does not implement in the disallowed_methods option. Note that
+; if your endpoint is truthful with its Allow header, then there is no need
+; to set this option. This option may be set in the general section or may
+; be set per endpoint. If this option is set both in the general section and
+; in a peer section, then the peer setting completely overrides the general
+; setting (i.e. the result is *not* the union of the two options).
+;
+; Note also that while Asterisk currently will parse an Allow header to learn
+; what methods an endpoint supports, the only actual use for this currently
+; is for determining if Asterisk may send connected line UPDATE requests and
+; MESSAGE requests. Its use may be expanded in the future.
+;
+; disallowed_methods = UPDATE
+
;
; Note that the TCP and TLS support for chan_sip is currently considered
; experimental. Since it is new, all of the related configuration options are
; Optionally add a port number, 192.168.1.1:5063 (default is port 5061)
; Remember that the IP address must match the common name (hostname) in the
; certificate, so you don't want to bind a TLS socket to multiple IP addresses.
+ ; For details how to construct a certificate for SIP see
+ ; http://tools.ietf.org/html/draft-ietf-sip-domain-certs
-;tlscertfile=</path/to/certificate.pem> ; Certificate file (*.pem only) to use for TLS connections
- ; default is to look for "asterisk.pem" in current directory
-
-;tlsprivatekey=</path/to/private.pem> ; Private key file (*.pem only) for TLS connections.
- ; If no tlsprivatekey is specified, tlscertfile is searched for
- ; for both public and private key.
-
-;tlscafile=</path/to/certificate>
-; If the server your connecting to uses a self signed certificate
-; you should have their certificate installed here so the code can
-; verify the authenticity of their certificate.
-
-;tlscadir=</path/to/ca/dir>
-; A directory full of CA certificates. The files must be named with
-; the CA subject name hash value.
-; (see man SSL_CTX_load_verify_locations for more info)
-
-;tlsdontverifyserver=[yes|no]
-; If set to yes, don't verify the servers certificate when acting as
-; a client. If you don't have the server's CA certificate you can
-; set this and it will connect without requiring tlscafile to be set.
-; Default is no.
+;tcpauthtimeout = 30 ; tcpauthtimeout specifies the maximum number
+ ; of seconds a client has to authenticate. If
+ ; the client does not authenticate beofre this
+ ; timeout expires, the client will be
+ ; disconnected. (default: 30 seconds)
-;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.
+;tcpauthlimit = 100 ; tcpauthlimit specifies the maximum number of
+ ; unauthenticated sessions that will be allowed
+ ; to connect at any given time. (default: 100)
srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; Note: Asterisk only uses the first host
; Disabling DNS SRV lookups disables the
; ability to place SIP calls based on domain
; names to some other SIP users on the Internet
+ ; Specifying a port in a SIP peer definition or
+ ; when dialing outbound calls will supress SRV
+ ; lookups for that peer or call.
;pedantic=yes ; Enable checking of tags in headers,
; international character conversions in URIs
; and multiline formatted headers for strict
- ; SIP compatibility (defaults to "no")
+ ; SIP compatibility (defaults to "yes")
-; See qos.tex or Quality of Service section of asterisk.pdf for a description of these parameters.
+; See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for a description of these parameters.
;tos_sip=cs3 ; Sets TOS for SIP packets.
;tos_audio=ef ; Sets TOS for RTP audio packets.
;tos_video=af41 ; Sets TOS for RTP video packets.
;minexpiry=60 ; Minimum length of registrations/subscriptions (default 60)
;defaultexpiry=120 ; Default length of incoming/outgoing registration
;mwiexpiry=3600 ; Expiry time for outgoing MWI subscriptions
-;qualifyfreq=60 ; Qualification: How often to check for the
- ; host to be up in seconds
- ; Set to low value if you use low timeout for
- ; NAT of UDP sessions
+;maxforwards=70 ; Setting for the SIP Max-Forwards: header (loop prevention)
+ ; Default value is 70
+;qualifyfreq=60 ; Qualification: How often to check for the host to be up in seconds
+ ; and reported in milliseconds with sip show settings.
+ ; Set to low value if you use low timeout for NAT of UDP sessions
+ ; Default: 60
;qualifygap=100 ; Number of milliseconds between each group of peers being qualified
+ ; Default: 100
;qualifypeers=1 ; Number of peers in a group to be qualified at the same time
+ ; Default: 1
;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY
;buggymwi=no ; Cisco SIP firmware doesn't support the MWI RFC
; fully. Enable this option to not get error messages
; when sending MWI to phones with this bug.
;mwi_from=asterisk ; When sending MWI NOTIFY requests, use this setting in
; the From: header as the "name" portion. Also fill the
- ; "user" portion of the URI in the From: header with this
- ; value if no fromuser is set
- ; Default: empty
+ ; "user" portion of the URI in the From: header with this
+ ; value if no fromuser is set
+ ; Default: empty
;vmexten=voicemail ; dialplan extension to reach mailbox sets the
; Message-Account in the MWI notify message
; defaults to "asterisk"
+; Codec negotiation
+;
+; When Asterisk is receiving a call, the codec will initially be set to the
+; first codec in the allowed codecs defined for the user receiving the call
+; that the caller also indicates that it supports. But, after the caller
+; starts sending RTP, Asterisk will switch to using whatever codec the caller
+; is sending.
+;
+; When Asterisk is placing a call, the codec used will be the first codec in
+; the allowed codecs that the callee indicates that it supports. Asterisk will
+; *not* switch to whatever codec the callee is sending.
+;
;preferred_codec_only=yes ; Respond to a SIP invite with the single most preferred codec
; rather than advertising all joint codec capabilities. This
; limits the other side's codec choice to exactly what we prefer.
;disallow=all ; First disallow all codecs
;allow=ulaw ; Allow codecs in order of preference
-;allow=ilbc ; see doc/rtp-packetization for framing options
+;allow=ilbc ; see https://wiki.asterisk.org/wiki/display/AST/RTP+Packetization
+ ; for framing options
;
; This option specifies a preference for which music on hold class this channel
; should listen to when put on hold if the music class has not been set on the
; Parkinglots are configured in features.conf
;language=en ; Default language setting for all users/peers
; This may also be set for individual users/peers
+;tonezone=se ; Default tonezone for all users/peers
+ ; This may also be set for individual users/peers
+
;relaxdtmf=yes ; Relax dtmf handling
;trustrpid = no ; If Remote-Party-ID should be trusted
-;sendrpid = yes ; If Remote-Party-ID should be sent
+;sendrpid = yes ; If Remote-Party-ID should be sent (defaults to no)
;sendrpid = rpid ; Use the "Remote-Party-ID" header
; to send the identity of the remote party
; This is identical to sendrpid=yes
; transmit such UPDATE messages to it, then you must enable this option.
; Otherwise, we will have to wait until we can send a reinvite to
; transmit the information.
+;prematuremedia=no ; Some ISDN links send empty media frames before
+ ; the call is in ringing or progress state. The SIP
+ ; channel will then send 183 indicating early media
+ ; which will be empty - thus users get no ring signal.
+ ; Setting this to "yes" will stop any media before we have
+ ; call progress (meaning the SIP channel will not send 183 Session
+ ; Progress for early media). Default is "yes". Also make sure that
+ ; the SIP peer is configured with progressinband=never.
+ ;
+ ; In order for "noanswer" applications to work, you need to run
+ ; the progress() application in the priority before the app.
;progressinband=never ; If we should generate in-band ringing always
; use 'never' to never use in-band signalling, even in cases
; The default user agent string also contains the Asterisk
; version. If you don't want to expose this, change the
; useragent string.
-;sdpsession=Asterisk PBX ; Allows you to change the SDP session name string, (s=)
- ; Like the useragent parameter, the default user agent string
- ; also contains the Asterisk version.
-;sdpowner=root ; Allows you to change the username field in the SDP owner string, (o=)
- ; This field MUST NOT contain spaces
;promiscredir = no ; If yes, allows 302 or REDIR to non-local SIP address
; Note that promiscredir when redirects are made to the
; local system will cause loops since Asterisk is incapable
; 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
;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
-;matchexterniplocally = yes ; Only substitute the externip or externhost setting if it matches
+;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.
; 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.
+
+; 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.
+;
+;------------------------ TLS settings ------------------------------------------------------------
+;tlscertfile=</path/to/certificate.pem> ; Certificate file (*.pem format only) to use for TLS connections
+ ; default is to look for "asterisk.pem" in current directory
+
+;tlsprivatekey=</path/to/private.pem> ; Private key file (*.pem 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.
;session-minse=90
;session-refresher=uas
;
-;--------------------------- HASH TABLE SIZES ------------------------------------------------
-; For maximum efficiency, adjust the following
-; values to be slightly larger than the maximum number of in-memory objects (devices).
-; Too large, and space is wasted. Too small, and things will run slower.
-; 563 is probably way too big for small (home) applications, but it
-; should cover most small/medium sites.
-; It is recommended to make the sizes be a prime number!
-; This was internally set to 17 for small-memory applications...
-; All tables default to 563, except when compiled in LOW_MEMORY mode,
-; in which case, they default to 17. You can override this by uncommenting
-; the following, and changing the values.
-;hash_users=563
-;hash_peers=563
-;hash_dialogs=563
-
;--------------------------- SIP DEBUGGING ---------------------------------------------------
;sipdebug = yes ; Turn on SIP debugging by default, from
; the moment the channel loads this configuration
;callcounter = yes ; Enable call counters on devices. This can be set per
; device too.
-;----------------------------------------- T.38 FAX PASSTHROUGH SUPPORT -----------------------
+;----------------------------------------- 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) passthrough on SIP to SIP calls, provided
-; both parties have T38 support enabled in their Asterisk configuration
-; This has to be enabled in the general section for all devices to work. You can then
-; disable it on a per device basis.
-;
-; T.38 faxing only works in SIP to SIP calls. It defaults to off.
+; 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.
;
-; Faxs Detect will cause the SIP channel to jump to the 'fax' extension (if it exists)
-; after T.38 is successfully negotiated.
-;
-; faxdetect = yes ; Default false
+; 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 => [transport://]user[@domain][:secret[:authuser]]@host[:port][/extension][~expiry]
+; register => [peer?][transport://]user[@domain][:secret[:authuser]]@host[:port][/extension][~expiry]
;
;
;
;
; 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' or 'tls'.
+; 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
+
;----------------------------------------- OUTBOUND MWI SUBSCRIPTIONS -------------------------
; Asterisk can subscribe to receive the MWI from another SIP server and store it locally for retrieval
-; by other phones.
+; 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]
+; 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
; to a host outside the NAT. This information is derived by one of the
; following (mutually exclusive) config file parameters:
;
-; a. "externip = hostname[:port]" specifies a static address[:port] to
+; 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 "bindport" value (which is
-; not guaranteed to work correctly, because a NAT box might remap the
+; 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:
;
-; externip = 12.34.56.78 ; use this address.
-; externip = 12.34.56.78:9900 ; use this address and port.
-; externip = mynat.my.org:12600 ; Public address of my nat box.
+; 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 "externip" except
+; 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.
; externhost=foo.dyndns.net ; refreshed periodically
; externrefresh=180 ; change the refresh interval
;
-; c. "stunaddr = stun.server[:port]" queries the STUN server specified
-; as an argument to obtain the external address/port.
-; Queries are also sent periodically every "externrefresh" seconds
-; (as a side effect, sending the query also acts as a keepalive for
-; the state entry on the nat box):
-;
-; stunaddr = foo.stun.com:3478
-; externrefresh = 15
-;
; Note that at the moment all these mechanism work only for the SIP socket.
-; The IP address discovered with externip/externhost/STUN is reused for
+; 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 "externip" and
-; "externhost" might not help you configure addresses properly, and you
-; really need to use STUN.
+; the internal<->external mapping. In these cases, the "externaddr" and
+; "externhost" might not help you configure addresses properly.
;
-; NOTE 2: when using "externip" or "externhost", the address part is
-; also used as the external address for media sessions.
-; If you use "stunaddr", STUN queries will be sent to the same server
-; also from media sockets, and this should permit a correct mapping of
-; the port numbers as well.
+; 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.
;
; nat = no ; Default. Use rport if the remote side says to use it.
; nat = force_rport ; Force rport to always be on.
-; nat = yes ; Force rport to always be on and perform symmetric RTP.
-; nat = comedia ; Use rport if the remote side says to use it and perform symmetric RTP.
+; nat = yes ; Force rport to always be on and perform comedia RTP handling.
+; nat = comedia ; Use rport if the remote side says to use it and perform comedia RTP handling.
+;
+; '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 port number they tell Asterisk to send RTP packets to
+; for their media streams is not actual port number 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 users 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
;----------------------------------- MEDIA HANDLING --------------------------------
-; By default, Asterisk tries to re-invite the audio to an optimal path. If there's
+; 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 with in the case where Asterisk is outside and have
-; clients on the inside of a NAT. In that case, you want to set canreinvite=nonat
+; 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.
;
-;canreinvite=yes ; Asterisk by default tries to redirect the
- ; RTP media stream (audio) to go directly from
+;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 wants Asterisk to
+ ; 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
; call directly between the endpoints instead of sending
; a re-INVITE).
-;directrtpsetup=yes ; Enable the new experimental direct RTP setup. This sets up
- ; the call directly with media peer-2-peer without re-invites.
- ; Will not work for video and cases where the callee sends
- ; RTP payloads and fmtp headers in the 200 OK that does not match the
- ; callers INVITE. This will also fail if canreinvite is enabled when
- ; the device is actually behind NAT.
+ ; 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.
-;canreinvite=nonat ; An additional option is to allow media path redirection
+;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).
-;canreinvite=update ; Yet a third option... use UPDATE for media path redirection,
+;directmedia=update ; Yet a third option... use UPDATE for media path redirection,
; instead of INVITE. This can be combined with 'nonat', as
- ; 'canreinvite=update,nonat'. It implies 'yes'.
+ ; 'directmedia=update,nonat'. It 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.
;ignoresdpversion=yes ; By default, Asterisk will honor the session version
; number in SDP packets and will only modify the SDP
; (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
+
;----------------------------------------- REALTIME SUPPORT ------------------------
; For additional information on ARA, the Asterisk Realtime Architecture,
-; please read realtime.txt and extconfig.txt in the /doc directory of the
-; source code.
+; 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
; 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
; (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".
+
+;----------------------------- SIP_CAUSE reporting ---------------------------------
+; storesipcause = no ; This option causes chan_sip to set the
+ ; HASH(SIP_CAUSE,<channel name>) channel variable
+ ; to the value of the last sip response.
+ ; WARNING: enabling this option carries a
+ ; significant performance burden. It should only
+ ; be used in low call volume situations. This
+ ; option defaults to "no".
+
;-----------------------------------------------------------------------------------
[authentication]
; Asterisk only matches on IP/port, not on names. This is mostly used for SIP
; trunks.
;
+; 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
; remotesecret
; transport
; dtmfmode
-; canreinvite
+; directmedia
; nat
; callgroup
; pickupgroup
; contactdeny ; is to register at the same IP as a SIP provider,
; ; then call oneself, and get redirected to that
; ; same location).
+; directmediapermit
+; directmediadeny
+; unsolicited_mailbox
+; use_q850_reason
+; maxforwards
+; encryption
+; description ; Used to provide a description of the peer in console output
;[sip_proxy]
; For incoming calls only. Example: FWD (Free World Dialup)
;transport=udp,tcp ; This sets the transport type to udp for outgoing, and will
; ; accept both tcp and udp. Default is udp. The first transport
; ; listed will always be used for outgoing connections.
+;unsolicited_mailbox=4015552299 ; If the remote SIP server sends an unsolicited MWI NOTIFY message the new/old
+; ; message count will be stored in the configured virtual mailbox. It can be used
+; ; by any device supporting MWI by specifying <configured value>@SIP_Remote as the
+; ; mailbox.
;
; Because you might have a large number of similar sections, it is generally
type=friend
[natted-phone](!,basic-options) ; another template inheriting basic-options
- nat=yes
- canreinvite=no
+ directmedia=no
host=dynamic
[public-phone](!,basic-options) ; another template inheriting basic-options
- nat=no
- canreinvite=yes
+ directmedia=yes
[my-codecs](!) ; a template for my preferred codecs
disallow=all
allow=gsm
allow=g723
allow=ulaw
+ ; Or, more simply:
+ ;allow=!all,ilbc,g729,gsm,g723,ulaw
[ulaw-phone](!) ; and another one for ulaw-only
disallow=all
allow=ulaw
+ ; Again, more simply:
+ ;allow=!all,ulaw
; and finally instantiate a few phones
;
;context=from-sip ; Where to start in the dialplan when this phone calls
;callerid=John Doe <1234> ; Full caller ID, to override the phones config
; on incoming calls to Asterisk
+;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
-;nat=no ; there is not NAT between phone and Asterisk
-;canreinvite=yes ; allow RTP voice traffic to bypass Asterisk
+;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)
;regexten=1234 ; When they register, create extension 1234
;callerid="Jane Smith" <5678>
;host=dynamic ; This device needs to register
-;nat=yes ; X-Lite is behind a NAT router
-;canreinvite=no ; Typically set to NO if behind NAT
+;directmedia=no ; Typically set to NO if behind NAT
;disallow=all
;allow=gsm ; GSM consumes far less bandwidth than ulaw
;allow=ulaw
;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.
;[cisco1]
;type=friend
;secret=blah
;qualify=200 ; Qualify peer is no more than 200ms away
-;nat=yes ; This phone may be natted
- ; Send SIP and RTP to the IP address that packet is
- ; received from instead of trusting SIP headers
;host=dynamic ; This device registers with us
-;canreinvite=no ; Asterisk by default tries to redirect the
+;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
projects / asterisk/asterisk.git / blobdiff