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Update sip.conf.sample documentation, reorganize a bit
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216694 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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@@ -141,40 +141,10 @@ tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0
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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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; Remember that the DNS entry for the common name (server name) in the
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; certificate must point to the IP address you bind to,
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; 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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;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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;tlscafile=</path/to/certificate>
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; If the server your connecting to uses a self signed certificate
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; you should have their certificate installed here so the code can
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; verify the authenticity of their certificate.
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;tlscadir=</path/to/ca/dir>
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; A directory full of CA certificates. The files must be named with
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; the CA subject name hash value.
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; (see man SSL_CTX_load_verify_locations for more info)
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;tlsdontverifyserver=[yes|no]
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; If set to yes, don't verify the servers certificate when acting as
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; a client. If you don't have the server's CA certificate you can
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; set this and it will connect without requiring tlscafile to be set.
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; Default is no.
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;tlscipher=<SSL cipher string>
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; A string specifying which SSL ciphers to use or not use
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; A list of valid SSL cipher strings can be found at:
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; http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS
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;
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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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srvlookup=yes ; Enable DNS SRV lookups on outbound calls
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; Note: Asterisk only uses the first host
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@@ -204,21 +174,22 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
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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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;qualifyfreq=60 ; Qualification: How often to check for the host to be up in seconds
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; Set to low value if you use low timeout for NAT of UDP sessions
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; Default: 60
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;qualifygap=100 ; Number of milliseconds between each group of peers being qualified
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; Default: 100
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;qualifypeers=1 ; Number of peers in a group to be qualified at the same time
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; Default: 1
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;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY
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;buggymwi=no ; Cisco SIP firmware doesn't support the MWI RFC
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; fully. Enable this option to not get error messages
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; when sending MWI to phones with this bug.
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;mwi_from=asterisk ; When sending MWI NOTIFY requests, use this setting in
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; the From: header as the "name" portion. Also fill the
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; "user" portion of the URI in the From: header with this
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; value if no fromuser is set
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; Default: empty
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; "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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@@ -253,7 +224,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
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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 = yes ; If Remote-Party-ID should be sent (defaults to no)
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;sendrpid = rpid ; Use the "Remote-Party-ID" header
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; to send the identity of the remote party
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; This is identical to sendrpid=yes
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@@ -280,11 +251,6 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
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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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;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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;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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@@ -368,6 +334,38 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
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; If you have qualify on and the peer becomes unreachable
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; this setting will enforce inactivation of the regexten
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; extension for the peer
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;------------------------ TLS settings ------------------------------------------------------------
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;tlscertfile=</path/to/certificate.pem> ; Certificate file (*.pem format only) to use for TLS connections
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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 format 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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;tlscafile=</path/to/certificate>
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; If the server your connecting to uses a self signed certificate
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; you should have their certificate installed here so the code can
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; verify the authenticity of their certificate.
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;tlscadir=</path/to/ca/dir>
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; A directory full of CA certificates. The files must be named with
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; the CA subject name hash value.
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; (see man SSL_CTX_load_verify_locations for more info)
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;tlsdontverifyserver=[yes|no]
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; If set to yes, don't verify the servers certificate when acting as
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; a client. If you don't have the server's CA certificate you can
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; set this and it will connect without requiring tlscafile to be set.
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; Default is no.
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;tlscipher=<SSL cipher string>
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; A string specifying which SSL ciphers to use or not use
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; A list of valid SSL cipher strings can be found at:
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; http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS
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;
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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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;
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;--------------------------- SIP timers ----------------------------------------------------
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; These timers are used primarily in INVITE transactions.
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@@ -420,6 +418,10 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
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;session-refresher=uas
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;
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;--------------------------- HASH TABLE SIZES ------------------------------------------------
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; Hash tables are used internally by the SIP driver to locate objects in memory.
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; For every incoming call, Asterisk will match properties of the call with in-memory
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; hash tables to locate a matching device, peer or user.
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;
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; For maximum efficiency, adjust the following
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; values to be slightly larger than the maximum number of in-memory objects (devices).
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; Too large, and space is wasted. Too small, and things will run slower.
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@@ -575,6 +577,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
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; 0 = continue forever, hammering the other server
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; until it accepts the registration
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; Default is 0 tries, continue forever
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;----------------------------------------- OUTBOUND MWI SUBSCRIPTIONS -------------------------
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; Asterisk can subscribe to receive the MWI from another SIP server and store it locally for retrieval
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; by other phones.
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@@ -692,13 +695,6 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
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; call directly between the endpoints instead of sending
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; a re-INVITE).
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;directrtpsetup=yes ; Enable the new experimental direct RTP setup. This sets up
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; the call directly with media peer-2-peer without re-invites.
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; Will not work for video and cases where the callee sends
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; RTP payloads and fmtp headers in the 200 OK that does not match the
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; callers INVITE. This will also fail if directmedia is enabled when
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; the device is actually behind NAT.
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;directmedia=nonat ; An additional option is to allow media path redirection
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; (reinvite) but only when the peer where the media is being
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; sent is known to not be behind a NAT (as the RTP core can
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@@ -709,6 +705,13 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
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; instead of INVITE. This can be combined with 'nonat', as
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; 'directmedia=update,nonat'. It implies 'yes'.
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;directrtpsetup=yes ; Enable the new experimental direct RTP setup. This sets up
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; the call directly with media peer-2-peer without re-invites.
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; Will not work for video and cases where the callee sends
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; RTP payloads and fmtp headers in the 200 OK that does not match the
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; callers INVITE. This will also fail if directmedia is enabled when
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; the device is actually behind NAT.
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;ignoresdpversion=yes ; By default, Asterisk will honor the session version
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; number in SDP packets and will only modify the SDP
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; session if the version number changes. This option will
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@@ -718,6 +721,12 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
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; (observed with Microsoft OCS). By default this option is
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; off.
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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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;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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;----------------------------------------- REALTIME SUPPORT ------------------------
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; For additional information on ARA, the Asterisk Realtime Architecture,
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; please read realtime.txt and extconfig.txt in the /doc directory of the
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