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Update sip.conf.sample documentation, reorganize a bit

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216694 65c4cc65-6c06-0410-ace0-fbb531ad65f3
certified/1.8.6
Olle Johansson 16 years ago
parent 109cab6862
commit 8af3a908a9
1 changed files with 62 additions and 53 deletions
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115
configs/sip.conf.sample
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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
;tlsenable=no ; Enable server for incoming TLS (secure) connections (default is no)
;tlsbindaddr=0.0.0.0 ; IP address for TLS server to bind to (0.0.0.0) binds to all interfaces)
; Optionally add a port number, 192.168.1.1:5063 (default is port 5061)
; Remember that the IP address must match the common name (hostname) in the
; certificate, so you don't want to bind a TLS socket to multiple IP addresses.
; Remember that the DNS entry for the common name (server name) in the
; certificate must point to the IP address you bind to,
; so you don't want to bind a TLS socket to multiple IP addresses.
;tlscertfile=</path/to/certificate.pem> ; Certificate file (*.pem only) to use for TLS connections
; default is to look for "asterisk.pem" in current directory
;tlsprivatekey=</path/to/private.pem> ; Private key file (*.pem only) for TLS connections.
; If no tlsprivatekey is specified, tlscertfile is searched for
; for both public and private key.
;tlscafile=</path/to/certificate>
; If the server your connecting to uses a self signed certificate
; you should have their certificate installed here so the code can
; verify the authenticity of their certificate.
;tlscadir=</path/to/ca/dir>
; A directory full of CA certificates. The files must be named with
; the CA subject name hash value.
; (see man SSL_CTX_load_verify_locations for more info)
;tlsdontverifyserver=[yes|no]
; If set to yes, don't verify the servers certificate when acting as
; a client. If you don't have the server's CA certificate you can
; set this and it will connect without requiring tlscafile to be set.
; Default is no.
;tlscipher=<SSL cipher string>
; A string specifying which SSL ciphers to use or not use
; A list of valid SSL cipher strings can be found at:
; http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS
;
;tlsclientmethod=tlsv1 ; values include tlsv1, sslv3, sslv2.
; Specify protocol for outbound client connections.
; If left unspecified, the default is sslv2.
srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; Note: Asterisk only uses the first host
@ -204,21 +174,22 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
;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
;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
; 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"
@ -253,7 +224,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; 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
@ -280,11 +251,6 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; 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
@ -368,6 +334,38 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; If you have qualify on and the peer becomes unreachable
; this setting will enforce inactivation of the regexten
; extension for the peer
;------------------------ 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.
;tlscadir=</path/to/ca/dir>
; A directory full of CA certificates. The files must be named with
; the CA subject name hash value.
; (see man SSL_CTX_load_verify_locations for more info)
;tlsdontverifyserver=[yes|no]
; If set to yes, don't verify the servers certificate when acting as
; a client. If you don't have the server's CA certificate you can
; set this and it will connect without requiring tlscafile to be set.
; Default is no.
;tlscipher=<SSL cipher string>
; A string specifying which SSL ciphers to use or not use
; A list of valid SSL cipher strings can be found at:
; http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS
;
;tlsclientmethod=tlsv1 ; values include tlsv1, sslv3, sslv2.
; Specify protocol for outbound client connections.
; If left unspecified, the default is sslv2.
;
;--------------------------- SIP timers ----------------------------------------------------
; These timers are used primarily in INVITE transactions.
@ -420,6 +418,10 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
;session-refresher=uas
;
;--------------------------- HASH TABLE SIZES ------------------------------------------------
; Hash tables are used internally by the SIP driver to locate objects in memory.
; For every incoming call, Asterisk will match properties of the call with in-memory
; hash tables to locate a matching device, peer or user.
;
; 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.
@ -575,6 +577,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; 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.
@ -692,13 +695,6 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; 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 directmedia is enabled when
; the device is actually behind NAT.
;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
@ -709,6 +705,13 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; instead of INVITE. This can be combined with 'nonat', as
; '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.
;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
@ -718,6 +721,12 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; (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
;----------------------------------------- REALTIME SUPPORT ------------------------
; For additional information on ARA, the Asterisk Realtime Architecture,
; please read realtime.txt and extconfig.txt in the /doc directory of the

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