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@ -88,18 +88,18 @@
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context=default ; Default context for incoming calls
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;allowguest=no ; Allow or reject guest calls (default is yes)
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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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; '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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;allowtransfer=no ; Disable all transfers (unless enabled in peers or users)
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; Default is enabled
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; Default is enabled
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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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; 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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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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; Optionally add a port number, 192.168.1.1:5062 (default is port 5060)
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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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@ -109,20 +109,20 @@ udpbindaddr=0.0.0.0 ; IP address to bind UDP listen socket to (0.0.0
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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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; 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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; 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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;tlscertfile=</path/to/certificate.pem> ; Certificate file (*.pem only) to use for TLS connections
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; default is to look for "asterisk.pem" in current directory
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; default is to look for "asterisk.pem" in current directory
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;tlsprivatekey=</path/to/private.pem> ; Private key file (*.pem only) for TLS connections.
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; If no tlsprivatekey is specified, tlscertfile is searched for
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; for both public and private key.
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; If no tlsprivatekey is specified, tlscertfile is searched for
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; for both public and private key.
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;tlscafile=</path/to/certificate>
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; If the server your connecting to uses a self signed certificate
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@ -146,20 +146,20 @@ tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0
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; http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS
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;
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;tlsclientmethod=tlsv1 ; values include tlsv1, sslv3, sslv2.
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; Specify protocol for outbound client connections.
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; If left unspecified, the default is sslv2.
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; Specify protocol for outbound client connections.
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; If left unspecified, the default is sslv2.
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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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; 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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;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 "no")
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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 "no")
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; See qos.tex or Quality of Service section of asterisk.pdf for a description of these parameters.
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;tos_sip=cs3 ; Sets TOS for SIP packets.
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@ -173,32 +173,32 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
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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
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; and subscriptions (seconds)
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; and subscriptions (seconds)
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;minexpiry=60 ; Minimum length of registrations/subscriptions (default 60)
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;defaultexpiry=120 ; Default length of incoming/outgoing registration
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;mwiexpiry=3600 ; Expiry time for outgoing MWI subscriptions
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;qualifyfreq=60 ; Qualification: How often to check for the
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; host to be up in seconds
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; Set to low value if you use low timeout for
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; NAT of UDP sessions
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; host to be up in seconds
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; Set to low value if you use low timeout for
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; NAT of UDP sessions
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;qualifygap=100 ; Number of milliseconds between each group of peers being qualified
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;qualifypeers=1 ; Number of peers in a group to be qualified at the same time
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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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; 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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; 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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; Message-Account in the MWI notify message
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; defaults to "asterisk"
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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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; 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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@ -220,83 +220,83 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
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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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; 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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; 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
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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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; 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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; 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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; 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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;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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; 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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; 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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;sdpsession=Asterisk PBX ; Allows you to change the SDP session name string, (s=)
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; Like the useragent parameter, the default user agent string
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; also contains the Asterisk version.
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; Like the useragent parameter, the default user agent string
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; also contains the Asterisk version.
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;sdpowner=root ; Allows you to change the username field in the SDP owner string, (o=)
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; This field MUST NOT contain spaces
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; This field MUST NOT contain spaces
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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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; 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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; 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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; 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
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; video support is enabled, but you can't enable it for
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; one peer only without enabling in the general section.
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; If you set videosupport to "always", then RTP ports will
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; always be set up for video, even on clients that don't
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; support it. This assists callfile-derived calls and
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; certain transferred calls to use always use video when
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; available. [yes|NO|always]
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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
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; video support is enabled, but you can't enable it for
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; one peer only without enabling in the general section.
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; If you set videosupport to "always", then RTP ports will
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; always be set up for video, even on clients that don't
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; support it. This assists callfile-derived calls and
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; certain transferred calls to use always use video when
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; available. [yes|NO|always]
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;maxcallbitrate=384 ; Maximum bitrate for video calls (default 384 kb/s)
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; Videosupport and maxcallbitrate is settable
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; for peers and users as well
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; Videosupport and maxcallbitrate is settable
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; for peers and users as well
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;callevents=no ; generate manager events when sip ua
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; performs events (e.g. hold)
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; performs events (e.g. hold)
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;authfailureevents=no ; generate manager "peerstatus" events when peer can't
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; authenticate with Asterisk. Peerstatus will be "rejected".
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; authenticate with Asterisk. Peerstatus will be "rejected".
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;alwaysauthreject = yes ; When an incoming INVITE or REGISTER is to be rejected,
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; for any reason, always reject with an identical response
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; equivalent to valid username and invalid password/hash
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; instead of letting the requester know whether there was
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; a matching user or peer for their request. This reduces
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; the ability of an attacker to scan for valid SIP usernames.
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; for any reason, always reject with an identical response
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; equivalent to valid username and invalid password/hash
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; instead of letting the requester know whether there was
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; a matching user or peer for their request. This reduces
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; the ability of an attacker to scan for valid SIP usernames.
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;g726nonstandard = yes ; If the peer negotiates G726-32 audio, use AAL2 packing
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; order instead of RFC3551 packing order (this is required
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; for Sipura and Grandstream ATAs, among others). This is
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; contrary to the RFC3551 specification, the peer _should_
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; be negotiating AAL2-G726-32 instead :-(
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; order instead of RFC3551 packing order (this is required
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; for Sipura and Grandstream ATAs, among others). This is
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; contrary to the RFC3551 specification, the peer _should_
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; be negotiating AAL2-G726-32 instead :-(
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;outboundproxy=proxy.provider.domain ; send outbound signaling to this proxy, not directly to the devices
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;outboundproxy=proxy.provider.domain:8080 ; send outbound signaling to this proxy, not directly to the devices
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;outboundproxy=proxy.provider.domain,force ; Send ALL outbound signalling to proxy, ignoring route: headers
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@ -304,18 +304,18 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
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; ; (could also be tcp,udp) - defining transports on the proxy line only
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; ; applies for the global proxy, otherwise use the transport= option
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;matchexterniplocally = yes ; Only substitute the externip or externhost setting if it matches
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; your localnet setting. Unless you have some sort of strange network
|
|
|
|
|
; setup you will not need to enable this.
|
|
|
|
|
; 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.
|
|
|
|
|
; 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.
|
|
|
|
|
; register their phones.
|
|
|
|
|
|
|
|
|
|
;engine=asterisk ; RTP engine to use when communicating with the device
|
|
|
|
|
|
|
|
|
|
@ -332,9 +332,9 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
|
|
|
|
|
;
|
|
|
|
|
;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
|
|
|
|
|
; If you have qualify on and the peer becomes unreachable
|
|
|
|
|
; this setting will enforce inactivation of the regexten
|
|
|
|
|
; extension for the peer
|
|
|
|
|
;
|
|
|
|
|
;--------------------------- SIP timers ----------------------------------------------------
|
|
|
|
|
; These timers are used primarily in INVITE transactions.
|
|
|
|
|
@ -342,13 +342,13 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
|
|
|
|
|
; 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
|
|
|
|
|
; Defaults to 100 ms
|
|
|
|
|
;timert1=500 ; Default T1 timer
|
|
|
|
|
; Defaults to 500 ms or the measured round-trip
|
|
|
|
|
; time to a peer (qualify=yes).
|
|
|
|
|
; 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
|
|
|
|
|
; 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
|
|
|
|
|
@ -356,15 +356,15 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
|
|
|
|
|
; 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.
|
|
|
|
|
; 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)
|
|
|
|
|
; 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)
|
|
|
|
|
; (default is off - zero)
|
|
|
|
|
|
|
|
|
|
;--------------------------- SIP Session-Timers (RFC 4028)------------------------------------
|
|
|
|
|
; SIP Session-Timers provide an end-to-end keep-alive mechanism for active SIP sessions.
|
|
|
|
|
@ -403,11 +403,11 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
|
|
|
|
|
|
|
|
|
|
;--------------------------- SIP DEBUGGING ---------------------------------------------------
|
|
|
|
|
;sipdebug = yes ; Turn on SIP debugging by default, from
|
|
|
|
|
; the moment the channel loads this configuration
|
|
|
|
|
; the moment the channel loads this configuration
|
|
|
|
|
;recordhistory=yes ; Record SIP history by default
|
|
|
|
|
; (see sip history / sip no history)
|
|
|
|
|
; (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
|
|
|
|
|
; SIP history is output to the DEBUG logging channel
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
;--------------------------- STATUS NOTIFICATIONS (SUBSCRIPTIONS) ----------------------------
|
|
|
|
|
@ -430,30 +430,30 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
|
|
|
|
|
;
|
|
|
|
|
;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
|
|
|
|
|
; 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)
|
|
|
|
|
; 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.
|
|
|
|
|
; 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.
|
|
|
|
|
; 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.
|
|
|
|
|
; device too.
|
|
|
|
|
|
|
|
|
|
;----------------------------------------- T.38 FAX PASSTHROUGH SUPPORT -----------------------
|
|
|
|
|
;
|
|
|
|
|
@ -536,9 +536,9 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
|
|
|
|
|
|
|
|
|
|
;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
|
|
|
|
|
; 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.
|
|
|
|
|
@ -645,43 +645,43 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
|
|
|
|
|
; clients on the inside of a NAT. In that case, you want to set canreinvite=nonat
|
|
|
|
|
;
|
|
|
|
|
;canreinvite=yes ; Asterisk by default tries to redirect the
|
|
|
|
|
; RTP media stream (audio) to go directly from
|
|
|
|
|
; the caller to the callee. Some devices do not
|
|
|
|
|
; support this (especially if one of them is behind a NAT).
|
|
|
|
|
; The default setting is YES. If you have all clients
|
|
|
|
|
; behind a NAT, or for some other reason wants Asterisk to
|
|
|
|
|
; stay in the audio path, you may want to turn this off.
|
|
|
|
|
|
|
|
|
|
; This setting also affect direct RTP
|
|
|
|
|
; at call setup (a new feature in 1.4 - setting up the
|
|
|
|
|
; call directly between the endpoints instead of sending
|
|
|
|
|
; a re-INVITE).
|
|
|
|
|
; RTP media stream (audio) to go directly from
|
|
|
|
|
; the caller to the callee. Some devices do not
|
|
|
|
|
; support this (especially if one of them is behind a NAT).
|
|
|
|
|
; The default setting is YES. If you have all clients
|
|
|
|
|
; behind a NAT, or for some other reason wants Asterisk to
|
|
|
|
|
; stay in the audio path, you may want to turn this off.
|
|
|
|
|
|
|
|
|
|
; This setting also affect direct RTP
|
|
|
|
|
; at call setup (a new feature in 1.4 - setting up the
|
|
|
|
|
; call directly between the endpoints instead of sending
|
|
|
|
|
; a re-INVITE).
|
|
|
|
|
|
|
|
|
|
;directrtpsetup=yes ; Enable the new experimental direct RTP setup. This sets up
|
|
|
|
|
; the call directly with media peer-2-peer without re-invites.
|
|
|
|
|
; Will not work for video and cases where the callee sends
|
|
|
|
|
; RTP payloads and fmtp headers in the 200 OK that does not match the
|
|
|
|
|
; callers INVITE. This will also fail if canreinvite is enabled when
|
|
|
|
|
; the device is actually behind NAT.
|
|
|
|
|
; the call directly with media peer-2-peer without re-invites.
|
|
|
|
|
; Will not work for video and cases where the callee sends
|
|
|
|
|
; RTP payloads and fmtp headers in the 200 OK that does not match the
|
|
|
|
|
; callers INVITE. This will also fail if canreinvite is enabled when
|
|
|
|
|
; the device is actually behind NAT.
|
|
|
|
|
|
|
|
|
|
;canreinvite=nonat ; An additional option is to allow media path redirection
|
|
|
|
|
; (reinvite) but only when the peer where the media is being
|
|
|
|
|
; sent is known to not be behind a NAT (as the RTP core can
|
|
|
|
|
; determine it based on the apparent IP address the media
|
|
|
|
|
; arrives from).
|
|
|
|
|
; (reinvite) but only when the peer where the media is being
|
|
|
|
|
; sent is known to not be behind a NAT (as the RTP core can
|
|
|
|
|
; determine it based on the apparent IP address the media
|
|
|
|
|
; arrives from).
|
|
|
|
|
|
|
|
|
|
;canreinvite=update ; Yet a third option... use UPDATE for media path redirection,
|
|
|
|
|
; instead of INVITE. This can be combined with 'nonat', as
|
|
|
|
|
; 'canreinvite=update,nonat'. It implies 'yes'.
|
|
|
|
|
; instead of INVITE. This can be combined with 'nonat', as
|
|
|
|
|
; 'canreinvite=update,nonat'. It implies 'yes'.
|
|
|
|
|
|
|
|
|
|
;ignoresdpversion=yes ; By default, Asterisk will honor the session version
|
|
|
|
|
; number in SDP packets and will only modify the SDP
|
|
|
|
|
; session if the version number changes. This option will
|
|
|
|
|
; force asterisk to ignore the SDP session version number
|
|
|
|
|
; and treat all SDP data as new data. This is required
|
|
|
|
|
; for devices that send us non standard SDP packets
|
|
|
|
|
; (observed with Microsoft OCS). By default this option is
|
|
|
|
|
; off.
|
|
|
|
|
; number in SDP packets and will only modify the SDP
|
|
|
|
|
; session if the version number changes. This option will
|
|
|
|
|
; force asterisk to ignore the SDP session version number
|
|
|
|
|
; and treat all SDP data as new data. This is required
|
|
|
|
|
; for devices that send us non standard SDP packets
|
|
|
|
|
; (observed with Microsoft OCS). By default this option is
|
|
|
|
|
; off.
|
|
|
|
|
|
|
|
|
|
;----------------------------------------- REALTIME SUPPORT ------------------------
|
|
|
|
|
; For additional information on ARA, the Asterisk Realtime Architecture,
|
|
|
|
|
@ -689,38 +689,38 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
|
|
|
|
|
; source code.
|
|
|
|
|
;
|
|
|
|
|
;rtcachefriends=yes ; Cache realtime friends by adding them to the internal list
|
|
|
|
|
; just like friends added from the config file only on a
|
|
|
|
|
; as-needed basis? (yes|no)
|
|
|
|
|
; 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
|
|
|
|
|
; 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.
|
|
|
|
|
; 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.
|
|
|
|
|
; 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
|
|
|
|
|
;
|
|
|
|
|
; 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 ------------------------
|
|
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|
|
; Incoming INVITE and REFER messages can be matched against a list of 'allowed'
|
|
|
|
|
@ -744,45 +744,45 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
|
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|
; allowexternaldomains=no
|
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|
|
|
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|
|
;domain=mydomain.tld,mydomain-incoming
|
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|
|
; Add domain and configure incoming context
|
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|
|
; for external calls to this domain
|
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|
|
; Add domain and configure incoming context
|
|
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|
|
; for external calls to this domain
|
|
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|
|
;domain=1.2.3.4 ; Add IP address as local domain
|
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|
|
; You can have several "domain" settings
|
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|
|
; You can have several "domain" settings
|
|
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|
|
;allowexternaldomains=no ; Disable INVITE and REFER to non-local domains
|
|
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|
|
; Default is yes
|
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|
|
; Default is yes
|
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|
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;autodomain=yes ; Turn this on to have Asterisk add local host
|
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|
|
; name and local IP to domain list.
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|
|
; name and local IP to domain list.
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|
|
; fromdomain=mydomain.tld ; When making outbound SIP INVITEs to
|
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|
|
; non-peers, use your primary domain "identity"
|
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|
|
; for From: headers instead of just your IP
|
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|
|
; address. This is to be polite and
|
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|
|
; it may be a mandatory requirement for some
|
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|
|
; destinations which do not have a prior
|
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|
|
|
; account relationship with your server.
|
|
|
|
|
; non-peers, use your primary domain "identity"
|
|
|
|
|
; for From: headers instead of just your IP
|
|
|
|
|
; address. This is to be polite and
|
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|
|
; it may be a mandatory requirement for some
|
|
|
|
|
; destinations which do not have a prior
|
|
|
|
|
; account relationship with your server.
|
|
|
|
|
|
|
|
|
|
;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
|
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|
|
; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
|
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|
|
; 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.
|
|
|
|
|
; 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".
|
|
|
|
|
; 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.
|
|
|
|
|
; 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.
|
|
|
|
|
; channel. Two implementations are currently available - "fixed"
|
|
|
|
|
; (with size always equals to jbmaxsize) and "adaptive" (with
|
|
|
|
|
; variable size, actually the new jb of IAX2). Defaults to fixed.
|
|
|
|
|
|
|
|
|
|
; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
|
|
|
|
|
;-----------------------------------------------------------------------------------
|
|
|
|
|
@ -919,7 +919,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
|
|
|
|
|
;busylevel=2 ; Signal busy at 2 or more calls
|
|
|
|
|
;outboundproxy=proxy.provider.domain ; send outbound signaling to this proxy, not directly to the peer
|
|
|
|
|
;port=80 ; The port number we want to connect to on the remote side
|
|
|
|
|
; Also used as "defaultport" in combination with "defaultip" settings
|
|
|
|
|
; Also used as "defaultport" in combination with "defaultip" settings
|
|
|
|
|
|
|
|
|
|
;--- sample definition for a provider
|
|
|
|
|
;[provider1]
|
|
|
|
|
@ -940,30 +940,30 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
|
|
|
|
|
; the templates uncommented as they will not harm:
|
|
|
|
|
|
|
|
|
|
[basic-options](!) ; a template
|
|
|
|
|
dtmfmode=rfc2833
|
|
|
|
|
context=from-office
|
|
|
|
|
type=friend
|
|
|
|
|
dtmfmode=rfc2833
|
|
|
|
|
context=from-office
|
|
|
|
|
type=friend
|
|
|
|
|
|
|
|
|
|
[natted-phone](!,basic-options) ; another template inheriting basic-options
|
|
|
|
|
nat=yes
|
|
|
|
|
canreinvite=no
|
|
|
|
|
host=dynamic
|
|
|
|
|
nat=yes
|
|
|
|
|
canreinvite=no
|
|
|
|
|
host=dynamic
|
|
|
|
|
|
|
|
|
|
[public-phone](!,basic-options) ; another template inheriting basic-options
|
|
|
|
|
nat=no
|
|
|
|
|
canreinvite=yes
|
|
|
|
|
nat=no
|
|
|
|
|
canreinvite=yes
|
|
|
|
|
|
|
|
|
|
[my-codecs](!) ; a template for my preferred codecs
|
|
|
|
|
disallow=all
|
|
|
|
|
allow=ilbc
|
|
|
|
|
allow=g729
|
|
|
|
|
allow=gsm
|
|
|
|
|
allow=g723
|
|
|
|
|
allow=ulaw
|
|
|
|
|
disallow=all
|
|
|
|
|
allow=ilbc
|
|
|
|
|
allow=g729
|
|
|
|
|
allow=gsm
|
|
|
|
|
allow=g723
|
|
|
|
|
allow=ulaw
|
|
|
|
|
|
|
|
|
|
[ulaw-phone](!) ; and another one for ulaw-only
|
|
|
|
|
disallow=all
|
|
|
|
|
allow=ulaw
|
|
|
|
|
disallow=all
|
|
|
|
|
allow=ulaw
|
|
|
|
|
|
|
|
|
|
; and finally instantiate a few phones
|
|
|
|
|
;
|
|
|
|
|
@ -982,31 +982,31 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
|
|
|
|
|
;type=friend
|
|
|
|
|
;context=from-sip ; Where to start in the dialplan when this phone calls
|
|
|
|
|
;callerid=John Doe <1234> ; Full caller ID, to override the phones config
|
|
|
|
|
; on incoming calls to Asterisk
|
|
|
|
|
; on incoming calls to Asterisk
|
|
|
|
|
;host=192.168.0.23 ; we have a static but private IP address
|
|
|
|
|
; No registration allowed
|
|
|
|
|
; No registration allowed
|
|
|
|
|
;nat=no ; there is not NAT between phone and Asterisk
|
|
|
|
|
;canreinvite=yes ; allow RTP voice traffic to bypass Asterisk
|
|
|
|
|
;dtmfmode=info ; either RFC2833 or INFO for the BudgeTone
|
|
|
|
|
;call-limit=1 ; permit only 1 outgoing call and 1 incoming call at a time
|
|
|
|
|
; from the phone to asterisk (deprecated)
|
|
|
|
|
; 1 for the explicit peer, 1 for the explicit user,
|
|
|
|
|
; remember that a friend equals 1 peer and 1 user in
|
|
|
|
|
; memory
|
|
|
|
|
; There is no combined call counter for a "friend"
|
|
|
|
|
; so there's currently no way in sip.conf to limit
|
|
|
|
|
; to one inbound or outbound call per phone. Use
|
|
|
|
|
; the group counters in the dial plan for that.
|
|
|
|
|
;
|
|
|
|
|
; 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!
|
|
|
|
|
; listed with allow= does NOT matter!
|
|
|
|
|
;allow=alaw
|
|
|
|
|
;allow=g723.1 ; Asterisk only supports g723.1 pass-thru!
|
|
|
|
|
;allow=g729 ; Pass-thru only unless g729 license obtained
|
|
|
|
|
;callingpres=allowed_passed_screen ; Set caller ID presentation
|
|
|
|
|
; See README.callingpres for more information
|
|
|
|
|
; See README.callingpres for more information
|
|
|
|
|
|
|
|
|
|
;[xlite1]
|
|
|
|
|
; Turn off silence suppression in X-Lite ("Transmit Silence"=YES)!
|
|
|
|
|
@ -1035,10 +1035,10 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
|
|
|
|
|
;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
|
|
|
|
|
; 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"
|
|
|
|
|
; 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!
|
|
|
|
|
|
|
|
|
|
@ -1051,7 +1051,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
|
|
|
|
|
;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
|
|
|
|
|
; 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"
|
|
|
|
|
@ -1062,16 +1062,16 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
|
|
|
|
|
;secret=blah
|
|
|
|
|
;host=dynamic
|
|
|
|
|
;insecure=port ; Allow matching of peer by IP address without
|
|
|
|
|
; matching port number
|
|
|
|
|
; 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
|
|
|
|
|
; 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
|
|
|
|
|
; 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
|
|
|
|
|
;
|
|
|
|
|
@ -1086,30 +1086,30 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
|
|
|
|
|
;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
|
|
|
|
|
; Send SIP and RTP to the IP address that packet is
|
|
|
|
|
; received from instead of trusting SIP headers
|
|
|
|
|
;host=dynamic ; This device registers with us
|
|
|
|
|
;canreinvite=no ; Asterisk by default tries to redirect the
|
|
|
|
|
; RTP media stream (audio) to go directly from
|
|
|
|
|
; the caller to the callee. Some devices do not
|
|
|
|
|
; support this (especially if one of them is
|
|
|
|
|
; behind a NAT).
|
|
|
|
|
; 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
|
|
|
|
|
; 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.
|
|
|
|
|
; 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.
|
|
|
|
|
; 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.
|
|
|
|
|
; 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.
|
|
|
|
|
|
Sean Bright
16 years ago