2002-06-16 16:06:38 +00:00
|
|
|
;
|
2005-01-09 18:05:41 +00:00
|
|
|
; SIP Configuration example for Asterisk
|
2002-06-16 16:06:38 +00:00
|
|
|
;
|
2009-11-12 10:22:30 +00:00
|
|
|
; 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.
|
2010-07-16 12:13:45 +00:00
|
|
|
;
|
2009-11-12 13:54:45 +00:00
|
|
|
; 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
|
2009-11-12 10:22:30 +00:00
|
|
|
;
|
2007-12-19 08:57:45 +00:00
|
|
|
; SIP dial strings
|
|
|
|
|
;-----------------------------------------------------------
|
2009-05-28 14:39:21 +00:00
|
|
|
; In the dialplan (extensions.conf) you can use several
|
2007-12-19 08:57:45 +00:00
|
|
|
; syntaxes for dialing SIP devices.
|
2008-08-14 14:15:50 +00:00
|
|
|
; SIP/devicename
|
|
|
|
|
; SIP/username@domain (SIP uri)
|
|
|
|
|
; SIP/username[:password[:md5secret[:authname[:transport]]]]@host[:port]
|
|
|
|
|
; SIP/devicename/extension
|
2010-07-16 12:13:45 +00:00
|
|
|
; SIP/devicename/extension/IPorHost
|
|
|
|
|
; SIP/username@domain//IPorHost
|
2004-03-19 20:30:03 +00:00
|
|
|
;
|
2007-12-19 08:57:45 +00:00
|
|
|
;
|
|
|
|
|
; Devicename
|
2008-08-14 14:15:50 +00:00
|
|
|
; devicename is defined as a peer in a section below.
|
2007-12-19 08:57:45 +00:00
|
|
|
;
|
|
|
|
|
; username@domain
|
2008-08-14 14:15:50 +00:00
|
|
|
; Call any SIP user on the Internet
|
|
|
|
|
; (Don't forget to enable DNS SRV records if you want to use this)
|
2009-05-28 14:39:21 +00:00
|
|
|
;
|
2007-12-19 08:57:45 +00:00
|
|
|
; devicename/extension
|
2008-08-14 14:15:50 +00:00
|
|
|
; If you define a SIP proxy as a peer below, you may call
|
2009-05-28 14:39:21 +00:00
|
|
|
; SIP/proxyhostname/user or SIP/user@proxyhostname
|
|
|
|
|
; where the proxyhostname is defined in a section below
|
2008-08-14 14:15:50 +00:00
|
|
|
; This syntax also works with ATA's with FXO ports
|
2007-12-19 08:57:45 +00:00
|
|
|
;
|
2008-02-29 18:34:46 +00:00
|
|
|
; SIP/username[:password[:md5secret[:authname]]]@host[:port]
|
2008-08-14 14:15:50 +00:00
|
|
|
; This form allows you to specify password or md5secret and authname
|
|
|
|
|
; without altering any authentication data in config.
|
|
|
|
|
; Examples:
|
2008-02-29 18:34:46 +00:00
|
|
|
;
|
2008-08-14 14:15:50 +00:00
|
|
|
; SIP/*98@mysipproxy
|
|
|
|
|
; SIP/sales:topsecret::account02@domain.com:5062
|
|
|
|
|
; SIP/12345678::bc53f0ba8ceb1ded2b70e05c3f91de4f:myname@192.168.0.1
|
2008-02-29 18:34:46 +00:00
|
|
|
;
|
2010-07-16 12:13:45 +00:00
|
|
|
; IPorHost
|
|
|
|
|
; The next server for this call regardless of domain/peer
|
|
|
|
|
;
|
2007-12-19 08:57:45 +00:00
|
|
|
; 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
|
|
|
|
|
;
|
2008-08-14 14:15:50 +00:00
|
|
|
; SIP/sales@mysipproxy!sales@edvina.net
|
2007-12-19 08:57:45 +00:00
|
|
|
;
|
2010-04-13 16:18:16 +00:00
|
|
|
; 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
|
|
|
|
|
;
|
2007-12-19 08:57:45 +00:00
|
|
|
; CLI Commands
|
|
|
|
|
; -------------------------------------------------------------
|
2004-03-19 20:30:03 +00:00
|
|
|
; Useful CLI commands to check peers/users:
|
2008-08-14 14:15:50 +00:00
|
|
|
; sip show peers Show all SIP peers (including friends)
|
|
|
|
|
; sip show registry Show status of hosts we register with
|
2004-03-19 20:30:03 +00:00
|
|
|
;
|
2009-03-11 23:14:22 +00:00
|
|
|
; sip set debug on Show all SIP messages
|
2004-03-19 20:30:03 +00:00
|
|
|
;
|
2010-07-16 12:13:45 +00:00
|
|
|
; sip reload Reload configuration file
|
|
|
|
|
; sip show settings Show the current channel configuration
|
2005-03-17 15:56:55 +00:00
|
|
|
;
|
2009-01-28 13:11:44 +00:00
|
|
|
;------- Naming devices ------------------------------------------------------
|
|
|
|
|
;
|
|
|
|
|
; When naming devices, make sure you understand how Asterisk matches calls
|
|
|
|
|
; that come in.
|
|
|
|
|
; 1. Asterisk checks the SIP From: address username and matches against
|
2009-05-28 14:39:21 +00:00
|
|
|
; names of devices with type=user
|
2009-01-28 13:11:44 +00:00
|
|
|
; The name is the text between square brackets [name]
|
2009-01-28 13:26:31 +00:00
|
|
|
; 2. Asterisk checks the From: addres and matches the list of devices
|
|
|
|
|
; with a type=peer
|
|
|
|
|
; 3. Asterisk checks the IP address (and port number) that the INVITE
|
2009-01-28 13:11:44 +00:00
|
|
|
; was sent from and matches against any devices with type=peer
|
|
|
|
|
;
|
|
|
|
|
; Don't mix extensions with the names of the devices. Devices need a unique
|
|
|
|
|
; name. The device name is *not* used as phone numbers. Phone numbers are
|
|
|
|
|
; anything you declare as an extension in the dialplan (extensions.conf).
|
2009-05-28 14:39:21 +00:00
|
|
|
;
|
2009-01-28 13:26:31 +00:00
|
|
|
; When setting up trunks, make sure there's no risk that any From: username
|
2009-05-28 14:39:21 +00:00
|
|
|
; (caller ID) will match any of your device names, because then Asterisk
|
2009-01-28 13:26:31 +00:00
|
|
|
; might match the wrong device.
|
2009-01-28 13:11:44 +00:00
|
|
|
;
|
|
|
|
|
; Note: The parameter "username" is not the username and in most cases is
|
|
|
|
|
; not needed at all. Check below. In later releases, it's renamed
|
2009-05-28 14:39:21 +00:00
|
|
|
; to "defaultuser" which is a better name, since it is used in
|
2009-01-28 13:11:44 +00:00
|
|
|
; combination with the "defaultip" setting.
|
|
|
|
|
;-----------------------------------------------------------------------------
|
2004-03-19 20:30:03 +00:00
|
|
|
|
2009-11-12 10:24:20 +00:00
|
|
|
; ** 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)
|
2007-11-25 11:46:17 +00:00
|
|
|
; You are encouraged to use the dialplan groupcount functionality
|
|
|
|
|
; to enforce call limits instead of using this channel-specific method.
|
2009-05-28 14:39:21 +00:00
|
|
|
; You can still set limits per device in sip.conf or in a database by using
|
2007-11-25 11:46:17 +00:00
|
|
|
; "setvar" to set variables that can be used in the dialplan for various limits.
|
|
|
|
|
|
2002-06-16 16:06:38 +00:00
|
|
|
[general]
|
2008-08-14 14:15:50 +00:00
|
|
|
context=default ; Default context for incoming calls
|
|
|
|
|
;allowguest=no ; Allow or reject guest calls (default is yes)
|
2009-11-12 10:22:30 +00:00
|
|
|
; 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).
|
2008-08-14 14:15:50 +00:00
|
|
|
;match_auth_username=yes ; if available, match user entry using the
|
2009-05-28 14:39:21 +00:00
|
|
|
; 'username' field from the authentication line
|
|
|
|
|
; instead of the From: field.
|
2008-08-14 14:15:50 +00:00
|
|
|
allowoverlap=no ; Disable overlap dialing support. (Default is yes)
|
|
|
|
|
;allowtransfer=no ; Disable all transfers (unless enabled in peers or users)
|
2010-03-31 17:48:09 +00:00
|
|
|
; Default is enabled. The Dial() options 't' and 'T' are not
|
|
|
|
|
; related as to whether SIP transfers are allowed or not.
|
2008-08-14 14:15:50 +00:00
|
|
|
;realm=mydomain.tld ; Realm for digest authentication
|
2009-05-28 14:39:21 +00:00
|
|
|
; 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
|
2009-08-12 22:18:09 +00:00
|
|
|
;domainsasrealm=no ; Use domans 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.
|
2010-08-03 19:58:59 +00:00
|
|
|
|
|
|
|
|
; 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".)
|
2010-08-09 14:32:59 +00:00
|
|
|
; (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.)
|
2010-08-03 19:58:59 +00:00
|
|
|
;
|
|
|
|
|
; 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.
|
|
|
|
|
|
2008-08-14 14:15:50 +00:00
|
|
|
udpbindaddr=0.0.0.0 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all)
|
2009-05-28 14:39:21 +00:00
|
|
|
; Optionally add a port number, 192.168.1.1:5062 (default is port 5060)
|
2008-01-18 22:04:33 +00:00
|
|
|
|
2009-07-31 17:57:00 +00:00
|
|
|
; 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. Its
|
|
|
|
|
; use may be expanded in the future.
|
|
|
|
|
;
|
|
|
|
|
; disallowed_methods = UPDATE
|
|
|
|
|
|
2008-03-21 15:24:43 +00:00
|
|
|
;
|
|
|
|
|
; 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
|
|
|
|
|
; subject to change in any release. If they are changed, the changes will
|
|
|
|
|
; be reflected in this sample configuration file, as well as in the UPGRADE.txt file.
|
|
|
|
|
;
|
2008-07-06 20:11:37 +00:00
|
|
|
tcpenable=no ; Enable server for incoming TCP connections (default is no)
|
2008-08-14 14:15:50 +00:00
|
|
|
tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces)
|
2009-05-28 14:39:21 +00:00
|
|
|
; Optionally add a port number, 192.168.1.1:5062 (default is port 5060)
|
2008-01-22 09:57:16 +00:00
|
|
|
|
|
|
|
|
;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)
|
2009-05-28 14:39:21 +00:00
|
|
|
; Optionally add a port number, 192.168.1.1:5063 (default is port 5061)
|
2010-03-26 19:27:56 +00:00
|
|
|
; 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.
|
2010-03-26 19:07:38 +00:00
|
|
|
; For details how to construct a certificate for SIP see
|
|
|
|
|
; http://tools.ietf.org/html/draft-ietf-sip-domain-certs
|
2008-08-14 14:15:50 +00:00
|
|
|
|
|
|
|
|
srvlookup=yes ; Enable DNS SRV lookups on outbound calls
|
2009-05-28 14:39:21 +00:00
|
|
|
; Note: Asterisk only uses the first host
|
|
|
|
|
; in SRV records
|
|
|
|
|
; Disabling DNS SRV lookups disables the
|
|
|
|
|
; ability to place SIP calls based on domain
|
|
|
|
|
; names to some other SIP users on the Internet
|
2009-09-30 20:40:20 +00:00
|
|
|
; Specifying a port in a SIP peer definition or
|
|
|
|
|
; when dialing outbound calls will supress SRV
|
|
|
|
|
; lookups for that peer or call.
|
2008-08-14 14:15:50 +00:00
|
|
|
|
2009-05-28 14:39:21 +00:00
|
|
|
;pedantic=yes ; Enable checking of tags in headers,
|
|
|
|
|
; international character conversions in URIs
|
|
|
|
|
; and multiline formatted headers for strict
|
2010-08-24 20:36:35 +00:00
|
|
|
; SIP compatibility (defaults to "yes")
|
2006-03-28 03:28:52 +00:00
|
|
|
|
2011-02-02 19:30:49 +00:00
|
|
|
; See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for a description of these parameters.
|
2006-03-28 03:28:52 +00:00
|
|
|
;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.
|
2007-04-30 16:16:26 +00:00
|
|
|
;tos_text=af41 ; Sets TOS for RTP text packets.
|
|
|
|
|
|
2007-12-16 10:51:53 +00:00
|
|
|
;cos_sip=3 ; Sets 802.1p priority for SIP packets.
|
|
|
|
|
;cos_audio=5 ; Sets 802.1p priority for RTP audio packets.
|
|
|
|
|
;cos_video=4 ; Sets 802.1p priority for RTP video packets.
|
|
|
|
|
;cos_text=3 ; Sets 802.1p priority for RTP text packets.
|
2006-03-28 03:28:52 +00:00
|
|
|
|
2008-08-14 14:15:50 +00:00
|
|
|
;maxexpiry=3600 ; Maximum allowed time of incoming registrations
|
2009-05-28 14:39:21 +00:00
|
|
|
; and subscriptions (seconds)
|
2008-08-14 14:15:50 +00:00
|
|
|
;minexpiry=60 ; Minimum length of registrations/subscriptions (default 60)
|
|
|
|
|
;defaultexpiry=120 ; Default length of incoming/outgoing registration
|
2008-10-09 01:40:49 +00:00
|
|
|
;mwiexpiry=3600 ; Expiry time for outgoing MWI subscriptions
|
2010-07-16 10:00:58 +00:00
|
|
|
;maxforwards=70 ; Setting for the SIP Max-Forwards: header (loop prevention)
|
|
|
|
|
; Default value is 70
|
2009-09-07 12:41:08 +00:00
|
|
|
;qualifyfreq=60 ; Qualification: How often to check for the host to be up in seconds
|
2011-02-01 21:16:31 +00:00
|
|
|
; and reported in milliseconds with sip show settings.
|
2009-09-07 12:41:08 +00:00
|
|
|
; Set to low value if you use low timeout for NAT of UDP sessions
|
|
|
|
|
; Default: 60
|
2008-12-16 20:47:31 +00:00
|
|
|
;qualifygap=100 ; Number of milliseconds between each group of peers being qualified
|
2009-09-07 12:41:08 +00:00
|
|
|
; Default: 100
|
2008-12-16 20:47:31 +00:00
|
|
|
;qualifypeers=1 ; Number of peers in a group to be qualified at the same time
|
2009-09-07 12:41:08 +00:00
|
|
|
; Default: 1
|
2008-08-14 14:15:50 +00:00
|
|
|
;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY
|
|
|
|
|
;buggymwi=no ; Cisco SIP firmware doesn't support the MWI RFC
|
2009-05-28 14:39:21 +00:00
|
|
|
; fully. Enable this option to not get error messages
|
|
|
|
|
; when sending MWI to phones with this bug.
|
2009-04-09 21:06:26 +00:00
|
|
|
;mwi_from=asterisk ; When sending MWI NOTIFY requests, use this setting in
|
2009-05-28 14:39:21 +00:00
|
|
|
; the From: header as the "name" portion. Also fill the
|
2009-09-07 12:41:08 +00:00
|
|
|
; "user" portion of the URI in the From: header with this
|
|
|
|
|
; value if no fromuser is set
|
|
|
|
|
; Default: empty
|
2009-05-28 14:39:21 +00:00
|
|
|
;vmexten=voicemail ; dialplan extension to reach mailbox sets the
|
|
|
|
|
; Message-Account in the MWI notify message
|
|
|
|
|
; defaults to "asterisk"
|
2009-03-24 20:01:29 +00:00
|
|
|
|
2010-08-19 02:20:42 +00:00
|
|
|
; 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.
|
|
|
|
|
;
|
2009-03-24 20:01:29 +00:00
|
|
|
;preferred_codec_only=yes ; Respond to a SIP invite with the single most preferred codec
|
2009-05-28 14:39:21 +00:00
|
|
|
; rather than advertising all joint codec capabilities. This
|
|
|
|
|
; limits the other side's codec choice to exactly what we prefer.
|
2009-03-24 20:01:29 +00:00
|
|
|
|
2008-08-14 14:15:50 +00:00
|
|
|
;disallow=all ; First disallow all codecs
|
|
|
|
|
;allow=ulaw ; Allow codecs in order of preference
|
2011-02-02 18:59:29 +00:00
|
|
|
;allow=ilbc ; see https://wiki.asterisk.org/wiki/display/AST/RTP+Packetization
|
|
|
|
|
; for framing options
|
2006-07-19 20:44:39 +00:00
|
|
|
;
|
|
|
|
|
; 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
|
|
|
|
|
; channel with Set(CHANNEL(musicclass)=whatever) in the dialplan, and the peer
|
|
|
|
|
; channel putting this one on hold did not suggest a music class.
|
|
|
|
|
;
|
|
|
|
|
; This option may be specified globally, or on a per-user or per-peer basis.
|
|
|
|
|
;
|
|
|
|
|
;mohinterpret=default
|
|
|
|
|
;
|
|
|
|
|
; This option specifies which music on hold class to suggest to the peer channel
|
|
|
|
|
; when this channel places the peer on hold. It may be specified globally or on
|
|
|
|
|
; a per-user or per-peer basis.
|
|
|
|
|
;
|
|
|
|
|
;mohsuggest=default
|
|
|
|
|
;
|
2008-08-14 14:15:50 +00:00
|
|
|
;parkinglot=plaza ; Sets the default parking lot for call parking
|
2009-05-28 14:39:21 +00:00
|
|
|
; This may also be set for individual users/peers
|
|
|
|
|
; Parkinglots are configured in features.conf
|
2008-08-14 14:15:50 +00:00
|
|
|
;language=en ; Default language setting for all users/peers
|
2009-05-28 14:39:21 +00:00
|
|
|
; This may also be set for individual users/peers
|
2008-08-14 14:15:50 +00:00
|
|
|
;relaxdtmf=yes ; Relax dtmf handling
|
|
|
|
|
;trustrpid = no ; If Remote-Party-ID should be trusted
|
2009-09-07 12:41:08 +00:00
|
|
|
;sendrpid = yes ; If Remote-Party-ID should be sent (defaults to no)
|
2009-04-03 22:41:46 +00:00
|
|
|
;sendrpid = rpid ; Use the "Remote-Party-ID" header
|
2009-05-28 14:39:21 +00:00
|
|
|
; to send the identity of the remote party
|
|
|
|
|
; This is identical to sendrpid=yes
|
2009-04-03 22:41:46 +00:00
|
|
|
;sendrpid = pai ; Use the "P-Asserted-Identity" header
|
2009-05-28 14:39:21 +00:00
|
|
|
; to send the identity of the remote party
|
Add basic support for handling connected line-related UPDATE requests.
SIP purists may want to look the other way...
When COLP/CONP support for SIP was committed, there was a condition under
which Asterisk may transmit a SIP UPDATE in order to communicate the change
in connected line information. The issue here is that while we could send a
SIP UPDATE message, we were not prepared to receive such an UPDATE and would
always responde with a 501 when we received an UPDATE.
The situation was a bit rough. We really want to be able to receive UPDATEs
having to do with connected line changes, but the amount of effort involved
in properly supporting RFC 3311 was staggering. This commit represents a
compromise.
First, it was decided that it is important to only send a SIP UPDATE to
an endpoint that is able to handle one. So, now we have added parsing of
the Allow header into SIP. We store the allowed methods on SIP peers so
that when we communicate with them, we already will know what we can and
cannot send to them. We will parse the peer's allowed methods when he registers
with us. If the peer is not the type to register with us, but the qualify option
is enabled, then we will use the response to the OPTIONS request we send
the peer to determine the peer's allowed methods. When the peer's registration
expires, or when qualify deems the peer to be unreachable, we clear the allowed
methods from the peer.
For an actual call, we will copy the peer's allowed methods to the sip_pvt
representing the call leg. If we are communicating with an endpoint which is
not a peer, then we will just parse the Allow header from the first message
we receive during the call and store the information in the sip_pvt.
If, during communication with a peer, we receive a 501 response, then we will
make sure to save the fact that we cannot use that method when communicating
with that peer.
Now, with all that infrastructure in place, the only actual place we use this
information currently is when attempting to send a connected line change using
an UPDATE request. If we cannot send the change immediately using an UPDATE,
we will set the SIP_NEEDREINVITE flag so that we can send a REINVITE as soon
as it is allowed.
The second part of the changes here is for Asterisk to accept UPDATE requests
that have connected line changes. Since we are not fully supporting RFC 3311,
Asterisk will NOT place the UPDATE method in Allow headers it sends. Instead,
if you are communicating with what you know to be another Asterisk box, you may
set the rpid_update parameter in sip.conf so that we will send UPDATEs to that
Asterisk box. When we send a connected line update, we set a custom header
called "X-Asterisk-rpid-update."
On the receiving end, if Asterisk receives an UPDATE that does not have the
"X-Asterisk-rpid-update" header present, then Asterisk will respond with a 501
since media-changing UPDATEs are not supported. We should never get such
UPDATEs, since as was stated earlier, Asterisk does not put UPDATE in its Allow
header. If the custom header is present in the received UPDATE, though, then we
will check the incoming request for connected line updates and queue the update
on the channel where the change occurred.
ABE-1840
ABE-1822
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@195589 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-19 20:59:38 +00:00
|
|
|
;rpid_update = no ; In certain cases, the only method by which a connected line
|
2009-05-28 14:39:21 +00:00
|
|
|
; change may be immediately transmitted is with a SIP UPDATE request.
|
|
|
|
|
; If communicating with another Asterisk server, and you wish to be able
|
|
|
|
|
; 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.
|
2010-07-06 22:23:35 +00:00
|
|
|
;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.
|
Add basic support for handling connected line-related UPDATE requests.
SIP purists may want to look the other way...
When COLP/CONP support for SIP was committed, there was a condition under
which Asterisk may transmit a SIP UPDATE in order to communicate the change
in connected line information. The issue here is that while we could send a
SIP UPDATE message, we were not prepared to receive such an UPDATE and would
always responde with a 501 when we received an UPDATE.
The situation was a bit rough. We really want to be able to receive UPDATEs
having to do with connected line changes, but the amount of effort involved
in properly supporting RFC 3311 was staggering. This commit represents a
compromise.
First, it was decided that it is important to only send a SIP UPDATE to
an endpoint that is able to handle one. So, now we have added parsing of
the Allow header into SIP. We store the allowed methods on SIP peers so
that when we communicate with them, we already will know what we can and
cannot send to them. We will parse the peer's allowed methods when he registers
with us. If the peer is not the type to register with us, but the qualify option
is enabled, then we will use the response to the OPTIONS request we send
the peer to determine the peer's allowed methods. When the peer's registration
expires, or when qualify deems the peer to be unreachable, we clear the allowed
methods from the peer.
For an actual call, we will copy the peer's allowed methods to the sip_pvt
representing the call leg. If we are communicating with an endpoint which is
not a peer, then we will just parse the Allow header from the first message
we receive during the call and store the information in the sip_pvt.
If, during communication with a peer, we receive a 501 response, then we will
make sure to save the fact that we cannot use that method when communicating
with that peer.
Now, with all that infrastructure in place, the only actual place we use this
information currently is when attempting to send a connected line change using
an UPDATE request. If we cannot send the change immediately using an UPDATE,
we will set the SIP_NEEDREINVITE flag so that we can send a REINVITE as soon
as it is allowed.
The second part of the changes here is for Asterisk to accept UPDATE requests
that have connected line changes. Since we are not fully supporting RFC 3311,
Asterisk will NOT place the UPDATE method in Allow headers it sends. Instead,
if you are communicating with what you know to be another Asterisk box, you may
set the rpid_update parameter in sip.conf so that we will send UPDATEs to that
Asterisk box. When we send a connected line update, we set a custom header
called "X-Asterisk-rpid-update."
On the receiving end, if Asterisk receives an UPDATE that does not have the
"X-Asterisk-rpid-update" header present, then Asterisk will respond with a 501
since media-changing UPDATEs are not supported. We should never get such
UPDATEs, since as was stated earlier, Asterisk does not put UPDATE in its Allow
header. If the custom header is present in the received UPDATE, though, then we
will check the incoming request for connected line updates and queue the update
on the channel where the change occurred.
ABE-1840
ABE-1822
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@195589 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-19 20:59:38 +00:00
|
|
|
|
2008-08-14 14:15:50 +00:00
|
|
|
;progressinband=never ; If we should generate in-band ringing always
|
2009-05-28 14:39:21 +00:00
|
|
|
; use 'never' to never use in-band signalling, even in cases
|
|
|
|
|
; where some buggy devices might not render it
|
|
|
|
|
; Valid values: yes, no, never Default: never
|
2008-08-14 14:15:50 +00:00
|
|
|
;useragent=Asterisk PBX ; Allows you to change the user agent string
|
2009-05-28 14:39:21 +00:00
|
|
|
; The default user agent string also contains the Asterisk
|
|
|
|
|
; version. If you don't want to expose this, change the
|
|
|
|
|
; useragent string.
|
2008-08-14 14:15:50 +00:00
|
|
|
;promiscredir = no ; If yes, allows 302 or REDIR to non-local SIP address
|
2009-05-28 14:39:21 +00:00
|
|
|
; Note that promiscredir when redirects are made to the
|
|
|
|
|
; local system will cause loops since Asterisk is incapable
|
|
|
|
|
; of performing a "hairpin" call.
|
2008-08-14 14:15:50 +00:00
|
|
|
;usereqphone = no ; If yes, ";user=phone" is added to uri that contains
|
2009-05-28 14:39:21 +00:00
|
|
|
; a valid phone number
|
2008-08-14 14:15:50 +00:00
|
|
|
;dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. Default: rfc2833
|
2009-05-28 14:39:21 +00:00
|
|
|
; Other options:
|
|
|
|
|
; info : SIP INFO messages (application/dtmf-relay)
|
|
|
|
|
; shortinfo : SIP INFO messages (application/dtmf)
|
|
|
|
|
; inband : Inband audio (requires 64 kbit codec -alaw, ulaw)
|
|
|
|
|
; auto : Use rfc2833 if offered, inband otherwise
|
2008-08-14 14:15:50 +00:00
|
|
|
|
|
|
|
|
;compactheaders = yes ; send compact sip headers.
|
|
|
|
|
;
|
|
|
|
|
;videosupport=yes ; Turn on support for SIP video. You need to turn this
|
2009-05-28 14:39:21 +00:00
|
|
|
; on in this section to get any video support at all.
|
|
|
|
|
; You can turn it off on a per peer basis if the general
|
|
|
|
|
; video support is enabled, but you can't enable it for
|
|
|
|
|
; one peer only without enabling in the general section.
|
|
|
|
|
; If you set videosupport to "always", then RTP ports will
|
|
|
|
|
; always be set up for video, even on clients that don't
|
|
|
|
|
; support it. This assists callfile-derived calls and
|
|
|
|
|
; certain transferred calls to use always use video when
|
|
|
|
|
; available. [yes|NO|always]
|
2008-08-14 14:15:50 +00:00
|
|
|
|
|
|
|
|
;maxcallbitrate=384 ; Maximum bitrate for video calls (default 384 kb/s)
|
2009-05-28 14:39:21 +00:00
|
|
|
; Videosupport and maxcallbitrate is settable
|
|
|
|
|
; for peers and users as well
|
|
|
|
|
;callevents=no ; generate manager events when sip ua
|
|
|
|
|
; performs events (e.g. hold)
|
2008-08-14 14:15:50 +00:00
|
|
|
;authfailureevents=no ; generate manager "peerstatus" events when peer can't
|
2009-05-28 14:39:21 +00:00
|
|
|
; authenticate with Asterisk. Peerstatus will be "rejected".
|
2008-08-14 14:15:50 +00:00
|
|
|
;alwaysauthreject = yes ; When an incoming INVITE or REGISTER is to be rejected,
|
2009-05-28 14:39:21 +00:00
|
|
|
; for any reason, always reject with an identical response
|
|
|
|
|
; equivalent to valid username and invalid password/hash
|
|
|
|
|
; instead of letting the requester know whether there was
|
|
|
|
|
; a matching user or peer for their request. This reduces
|
|
|
|
|
; the ability of an attacker to scan for valid SIP usernames.
|
2010-08-10 21:50:24 +00:00
|
|
|
; This option is set to "yes" by default.
|
2008-08-14 14:15:50 +00:00
|
|
|
|
2010-09-03 22:23:47 +00:00
|
|
|
;auth_options_requests = yes ; Enabling this option will authenticate OPTIONS requests just like
|
|
|
|
|
; INVITE requests are. By default this option is disabled.
|
2010-09-03 17:30:04 +00:00
|
|
|
|
2008-08-14 14:15:50 +00:00
|
|
|
;g726nonstandard = yes ; If the peer negotiates G726-32 audio, use AAL2 packing
|
2009-05-28 14:39:21 +00:00
|
|
|
; order instead of RFC3551 packing order (this is required
|
|
|
|
|
; for Sipura and Grandstream ATAs, among others). This is
|
|
|
|
|
; contrary to the RFC3551 specification, the peer _should_
|
|
|
|
|
; be negotiating AAL2-G726-32 instead :-(
|
2008-08-14 14:15:50 +00:00
|
|
|
;outboundproxy=proxy.provider.domain ; send outbound signaling to this proxy, not directly to the devices
|
2007-02-11 19:42:55 +00:00
|
|
|
;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
|
2009-05-28 14:39:21 +00:00
|
|
|
;outboundproxy=tls://proxy.provider.domain ; same as '=proxy.provider.domain' except we try to connect with tls
|
2010-08-11 13:31:39 +00:00
|
|
|
;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
|
2008-07-01 21:03:52 +00:00
|
|
|
; ; (could also be tcp,udp) - defining transports on the proxy line only
|
|
|
|
|
; ; applies for the global proxy, otherwise use the transport= option
|
2010-07-19 17:16:23 +00:00
|
|
|
;matchexternaddrlocally = yes ; Only substitute the externaddr or externhost setting if it matches
|
2009-05-28 14:39:21 +00:00
|
|
|
; your localnet setting. Unless you have some sort of strange network
|
|
|
|
|
; setup you will not need to enable this.
|
2007-02-11 19:42:55 +00:00
|
|
|
|
2008-09-12 20:49:46 +00:00
|
|
|
;dynamic_exclude_static = yes ; Disallow all dynamic hosts from registering
|
2009-05-28 14:39:21 +00:00
|
|
|
; 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.
|
2008-09-12 20:49:46 +00:00
|
|
|
|
|
|
|
|
;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
|
2009-05-28 14:39:21 +00:00
|
|
|
; register their phones.
|
2008-09-12 20:49:46 +00:00
|
|
|
|
2009-04-02 17:20:52 +00:00
|
|
|
;engine=asterisk ; RTP engine to use when communicating with the device
|
|
|
|
|
|
2004-09-07 23:45:34 +00:00
|
|
|
;
|
2005-10-04 22:51:59 +00:00
|
|
|
; If regcontext is specified, Asterisk will dynamically create and destroy a
|
|
|
|
|
; NoOp priority 1 extension for a given peer who registers or unregisters with
|
2009-05-28 14:39:21 +00:00
|
|
|
; us and have a "regexten=" configuration item.
|
|
|
|
|
; Multiple contexts may be specified by separating them with '&'. The
|
2006-05-18 14:07:46 +00:00
|
|
|
; actual extension is the 'regexten' parameter of the registering peer or its
|
|
|
|
|
; name if 'regexten' is not provided. If more than one context is provided,
|
|
|
|
|
; the context must be specified within regexten by appending the desired
|
2009-05-28 14:39:21 +00:00
|
|
|
; context after '@'. More than one regexten may be supplied if they are
|
2006-05-18 14:07:46 +00:00
|
|
|
; separated by '&'. Patterns may be used in regexten.
|
2004-09-07 23:45:34 +00:00
|
|
|
;
|
2004-11-12 03:57:39 +00:00
|
|
|
;regcontext=sipregistrations
|
2008-08-14 14:15:50 +00:00
|
|
|
;regextenonqualify=yes ; Default "no"
|
2009-05-28 14:39:21 +00:00
|
|
|
; If you have qualify on and the peer becomes unreachable
|
|
|
|
|
; this setting will enforce inactivation of the regexten
|
|
|
|
|
; extension for the peer
|
2009-10-21 14:39:10 +00:00
|
|
|
|
|
|
|
|
; 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
|
|
|
|
|
|
2009-11-02 14:57:11 +00:00
|
|
|
|
|
|
|
|
;use_q850_reason = no ; Default "no"
|
|
|
|
|
; Set to yes add Reason header and use Reason header if it is available.
|
|
|
|
|
;
|
2009-09-07 12:41:08 +00:00
|
|
|
;------------------------ 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.
|
|
|
|
|
|
2011-01-31 22:26:06 +00:00
|
|
|
;tlscapath=</path/to/ca/dir>
|
2009-09-07 12:41:08 +00:00
|
|
|
; 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.
|
2006-06-29 07:04:43 +00:00
|
|
|
;
|
2007-12-16 08:19:38 +00:00
|
|
|
;--------------------------- SIP timers ----------------------------------------------------
|
2009-05-28 14:39:21 +00:00
|
|
|
; These timers are used primarily in INVITE transactions.
|
2007-12-16 08:19:38 +00:00
|
|
|
; The default for Timer T1 is 500 ms or the measured run-trip time between
|
|
|
|
|
; Asterisk and the device if you have qualify=yes for the device.
|
|
|
|
|
;
|
2008-08-14 14:15:50 +00:00
|
|
|
;t1min=100 ; Minimum roundtrip time for messages to monitored hosts
|
2009-05-28 14:39:21 +00:00
|
|
|
; Defaults to 100 ms
|
2008-08-14 14:15:50 +00:00
|
|
|
;timert1=500 ; Default T1 timer
|
2009-05-28 14:39:21 +00:00
|
|
|
; Defaults to 500 ms or the measured round-trip
|
|
|
|
|
; time to a peer (qualify=yes).
|
2008-08-14 14:15:50 +00:00
|
|
|
;timerb=32000 ; Call setup timer. If a provisional response is not received
|
2009-05-28 14:39:21 +00:00
|
|
|
; in this amount of time, the call will autocongest
|
|
|
|
|
; Defaults to 64*timert1
|
2007-12-16 08:19:38 +00:00
|
|
|
|
2006-12-02 12:05:40 +00:00
|
|
|
;--------------------------- RTP timers ----------------------------------------------------
|
|
|
|
|
; These timers are currently used for both audio and video streams. The RTP timeouts
|
|
|
|
|
; are only applied to the audio channel.
|
|
|
|
|
; The settings are settable in the global section as well as per device
|
|
|
|
|
;
|
2008-08-14 14:15:50 +00:00
|
|
|
;rtptimeout=60 ; Terminate call if 60 seconds of no RTP or RTCP activity
|
2009-05-28 14:39:21 +00:00
|
|
|
; 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.
|
2008-08-14 14:15:50 +00:00
|
|
|
;rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP or RTCP activity
|
2009-05-28 14:39:21 +00:00
|
|
|
; on the audio channel
|
|
|
|
|
; when we're on hold (must be > rtptimeout)
|
2008-08-14 14:15:50 +00:00
|
|
|
;rtpkeepalive=<secs> ; Send keepalives in the RTP stream to keep NAT open
|
2009-05-28 14:39:21 +00:00
|
|
|
; (default is off - zero)
|
2008-01-16 21:53:10 +00:00
|
|
|
|
|
|
|
|
;--------------------------- SIP Session-Timers (RFC 4028)------------------------------------
|
|
|
|
|
; SIP Session-Timers provide an end-to-end keep-alive mechanism for active SIP sessions.
|
|
|
|
|
; This mechanism can detect and reclaim SIP channels that do not terminate through normal
|
|
|
|
|
; signaling procedures. Session-Timers can be configured globally or at a user/peer level.
|
|
|
|
|
; The operation of Session-Timers is driven by the following configuration parameters:
|
|
|
|
|
;
|
|
|
|
|
; * session-timers - Session-Timers feature operates in the following three modes:
|
|
|
|
|
; originate : Request and run session-timers always
|
|
|
|
|
; accept : Run session-timers only when requested by other UA
|
|
|
|
|
; refuse : Do not run session timers in any case
|
|
|
|
|
; The default mode of operation is 'accept'.
|
|
|
|
|
; * session-expires - Maximum session refresh interval in seconds. Defaults to 1800 secs.
|
|
|
|
|
; * session-minse - Minimum session refresh interval in seconds. Defualts to 90 secs.
|
|
|
|
|
; * session-refresher - The session refresher (uac|uas). Defaults to 'uas'.
|
|
|
|
|
;
|
|
|
|
|
;session-timers=originate
|
|
|
|
|
;session-expires=600
|
|
|
|
|
;session-minse=90
|
|
|
|
|
;session-refresher=uas
|
2008-07-05 20:37:53 +00:00
|
|
|
;
|
2006-06-29 07:04:43 +00:00
|
|
|
;--------------------------- SIP DEBUGGING ---------------------------------------------------
|
2008-08-14 14:15:50 +00:00
|
|
|
;sipdebug = yes ; Turn on SIP debugging by default, from
|
2009-05-28 14:39:21 +00:00
|
|
|
; the moment the channel loads this configuration
|
|
|
|
|
;recordhistory=yes ; Record SIP history by default
|
|
|
|
|
; (see sip history / sip no history)
|
2008-08-14 14:15:50 +00:00
|
|
|
;dumphistory=yes ; Dump SIP history at end of SIP dialogue
|
2009-05-28 14:39:21 +00:00
|
|
|
; SIP history is output to the DEBUG logging channel
|
2006-06-29 07:04:43 +00:00
|
|
|
|
|
|
|
|
|
|
|
|
|
;--------------------------- STATUS NOTIFICATIONS (SUBSCRIPTIONS) ----------------------------
|
|
|
|
|
; You can subscribe to the status of extensions with a "hint" priority
|
|
|
|
|
; (See extensions.conf.sample for examples)
|
2009-05-28 14:39:21 +00:00
|
|
|
; chan_sip support two major formats for notifications: dialog-info and SIMPLE
|
2006-11-29 20:57:48 +00:00
|
|
|
;
|
2007-11-25 11:46:17 +00:00
|
|
|
; You will get more detailed reports (busy etc) if you have a call counter enabled
|
2009-05-28 14:39:21 +00:00
|
|
|
; for a device.
|
2006-11-29 20:57:48 +00:00
|
|
|
;
|
2009-05-28 14:39:21 +00:00
|
|
|
; If you set the busylevel, we will indicate busy when we have a number of calls that
|
2007-11-25 11:46:17 +00:00
|
|
|
; matches the busylevel treshold.
|
2007-02-01 20:43:49 +00:00
|
|
|
;
|
2006-11-29 20:57:48 +00:00
|
|
|
; For queues, you will need this level of detail in status reporting, regardless
|
|
|
|
|
; if you use SIP subscriptions. Queues and manager use the same internal interface
|
|
|
|
|
; for reading status information.
|
|
|
|
|
;
|
2006-06-29 07:04:43 +00:00
|
|
|
; Note: Subscriptions does not work if you have a realtime dialplan and use the
|
|
|
|
|
; realtime switch.
|
|
|
|
|
;
|
2008-08-14 14:15:50 +00:00
|
|
|
;allowsubscribe=no ; Disable support for subscriptions. (Default is yes)
|
|
|
|
|
;subscribecontext = default ; Set a specific context for SUBSCRIBE requests
|
2009-05-28 14:39:21 +00:00
|
|
|
; Useful to limit subscriptions to local extensions
|
|
|
|
|
; Settable per peer/user also
|
2008-11-01 01:55:04 +00:00
|
|
|
;notifyringing = no ; Control whether subscriptions already INUSE get sent
|
2009-05-28 14:39:21 +00:00
|
|
|
; RINGING when another call is sent (default: yes)
|
2008-08-14 14:15:50 +00:00
|
|
|
;notifyhold = yes ; Notify subscriptions on HOLD state (default: no)
|
2009-05-28 14:39:21 +00:00
|
|
|
; Turning on notifyringing and notifyhold will add a lot
|
|
|
|
|
; more database transactions if you are using realtime.
|
2008-11-04 17:00:45 +00:00
|
|
|
;notifycid = yes ; Control whether caller ID information is sent along with
|
2009-05-28 14:39:21 +00:00
|
|
|
; 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.
|
2008-08-14 14:15:50 +00:00
|
|
|
;callcounter = yes ; Enable call counters on devices. This can be set per
|
2009-05-28 14:39:21 +00:00
|
|
|
; device too.
|
2006-10-16 20:08:23 +00:00
|
|
|
|
2009-10-05 19:45:00 +00:00
|
|
|
;----------------------------------------- T.38 FAX SUPPORT ----------------------------------
|
2006-06-13 19:38:41 +00:00
|
|
|
;
|
2006-06-30 07:18:30 +00:00
|
|
|
; This setting is available in the [general] section as well as in device configurations.
|
2009-10-05 19:45:00 +00:00
|
|
|
; Setting this to yes enables T.38 FAX (UDPTL) on SIP calls; it defaults to off.
|
2006-12-05 16:48:15 +00:00
|
|
|
;
|
2009-06-26 19:27:24 +00:00
|
|
|
; 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.
|
2004-09-07 23:45:34 +00:00
|
|
|
;
|
2009-10-05 19:45:00 +00:00
|
|
|
; 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)
|
Improve handling of T.38 re-INVITEs that arrive before a T.38-capable
application is executing on a channel.
This patch addresses an issue found during working with end-users
using res_fax. If an incoming call is answered in the dialplan, or
jumps to the 'fax' extension due to reception of a CNG tone (with
faxdetect enabled), and then the remote endpoint sends a T.38
re-INVITE, it is possible for the channel's T.38 state to be
'T38_STATE_NEGOTIATING' when the application starts up. Unfortunately,
even if the application wants to use T.38, it can't respond to the
peer's negotiation request, because the AST_CONTROL_T38_PARAMETERS
control frame that chan_sip sent originally has been lost, and the
application needs the content of that frame to be able to formulate a
reply.
This patch adds a new 'request' type to AST_CONTROL_T38_PARAMETERS,
AST_T38_REQUEST_PARMS. If the application sends this request, chan_sip
will re-send the original control frame (with
AST_T38_REQUEST_NEGOTIATE as the request type), and the application
can respond as normal. If this occurs within the five second timeout
in chan_sip, the automatic cancellation of the peer reinvite will be
stopped, and the application will 'own' the negotiation process from
that point onwards.
This also improves the code path in chan_sip to allow sip_indicate(),
when called for AST_CONTROL_T38_PARAMETERS, to be able to return a
non-zero response, which should have been in place before since the
control frame *can* fail to be processed properly. It also modifies
ast_indicate() to return whatever result the channel driver returned
for this control frame, rather than converting all non-zero results
into '-1'. Finally, the new request type intentionally returns a
positive value, so that an application that sends
AST_T38_REQUEST_PARMS can know for certain whether the channel driver
accepted it and will be replying with a control frame of its own, or
whether it was ignored (if the sip_indicate()/ast_indicate() path had
properly supported failure responses before, this would not be
necessary).
This patch also modifies res_fax to take advantage of the new request.
In addition, this patch makes sip_t38_abort() actually lock the
private structure before doing its work... bad programmer, no donut.
This patch also enhances chan_sip's 'faxdetect' support to allow
triggering on T.38 re-INVITEs received as well as CNG tone detection.
Review: https://reviewboard.asterisk.org/r/556/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@254450 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-25 15:27:31 +00:00
|
|
|
; 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.
|
2009-05-28 14:39:21 +00:00
|
|
|
;
|
Improve handling of T.38 re-INVITEs that arrive before a T.38-capable
application is executing on a channel.
This patch addresses an issue found during working with end-users
using res_fax. If an incoming call is answered in the dialplan, or
jumps to the 'fax' extension due to reception of a CNG tone (with
faxdetect enabled), and then the remote endpoint sends a T.38
re-INVITE, it is possible for the channel's T.38 state to be
'T38_STATE_NEGOTIATING' when the application starts up. Unfortunately,
even if the application wants to use T.38, it can't respond to the
peer's negotiation request, because the AST_CONTROL_T38_PARAMETERS
control frame that chan_sip sent originally has been lost, and the
application needs the content of that frame to be able to formulate a
reply.
This patch adds a new 'request' type to AST_CONTROL_T38_PARAMETERS,
AST_T38_REQUEST_PARMS. If the application sends this request, chan_sip
will re-send the original control frame (with
AST_T38_REQUEST_NEGOTIATE as the request type), and the application
can respond as normal. If this occurs within the five second timeout
in chan_sip, the automatic cancellation of the peer reinvite will be
stopped, and the application will 'own' the negotiation process from
that point onwards.
This also improves the code path in chan_sip to allow sip_indicate(),
when called for AST_CONTROL_T38_PARAMETERS, to be able to return a
non-zero response, which should have been in place before since the
control frame *can* fail to be processed properly. It also modifies
ast_indicate() to return whatever result the channel driver returned
for this control frame, rather than converting all non-zero results
into '-1'. Finally, the new request type intentionally returns a
positive value, so that an application that sends
AST_T38_REQUEST_PARMS can know for certain whether the channel driver
accepted it and will be replying with a control frame of its own, or
whether it was ignored (if the sip_indicate()/ast_indicate() path had
properly supported failure responses before, this would not be
necessary).
This patch also modifies res_fax to take advantage of the new request.
In addition, this patch makes sip_t38_abort() actually lock the
private structure before doing its work... bad programmer, no donut.
This patch also enhances chan_sip's 'faxdetect' support to allow
triggering on T.38 re-INVITEs received as well as CNG tone detection.
Review: https://reviewboard.asterisk.org/r/556/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@254450 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-25 15:27:31 +00:00
|
|
|
; 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
|
2008-12-04 23:00:30 +00:00
|
|
|
;
|
2006-02-01 13:23:59 +00:00
|
|
|
;----------------------------------------- OUTBOUND SIP REGISTRATIONS ------------------------
|
2004-03-25 20:52:57 +00:00
|
|
|
; Asterisk can register as a SIP user agent to a SIP proxy (provider)
|
|
|
|
|
; Format for the register statement is:
|
2009-08-19 21:05:17 +00:00
|
|
|
; register => [peer?][transport://]user[@domain][:secret[:authuser]]@host[:port][/extension][~expiry]
|
2008-01-18 22:04:33 +00:00
|
|
|
;
|
2004-03-25 20:52:57 +00:00
|
|
|
;
|
2009-05-28 14:39:21 +00:00
|
|
|
;
|
|
|
|
|
; domain is either
|
2009-01-15 17:55:53 +00:00
|
|
|
; - domain in DNS
|
|
|
|
|
; - host name in DNS
|
|
|
|
|
; - the name of a peer defined below or in realtime
|
2009-05-28 14:39:21 +00:00
|
|
|
; The domain is where you register your username, so your SIP uri you are registering to
|
2009-01-15 17:55:53 +00:00
|
|
|
; is username@domain
|
|
|
|
|
;
|
2005-10-04 22:51:59 +00:00
|
|
|
; If no extension is given, the 's' extension is used. The extension needs to
|
|
|
|
|
; be defined in extensions.conf to be able to accept calls from this SIP proxy
|
|
|
|
|
; (provider).
|
2004-03-25 20:52:57 +00:00
|
|
|
;
|
2006-10-30 19:56:14 +00:00
|
|
|
; A similar effect can be achieved by adding a "callbackextension" option in a peer section.
|
2006-10-06 16:20:42 +00:00
|
|
|
; this is equivalent to having the following line in the general section:
|
|
|
|
|
;
|
2008-08-14 14:15:50 +00:00
|
|
|
; register => username:secret@host/callbackextension
|
2006-10-06 16:20:42 +00:00
|
|
|
;
|
|
|
|
|
; and more readable because you don't have to write the parameters in two places
|
|
|
|
|
; (note that the "port" is ignored - this is a bug that should be fixed).
|
|
|
|
|
;
|
2009-01-28 13:26:31 +00:00
|
|
|
; Note that a register= line doesn't mean that we will match the incoming call in any
|
|
|
|
|
; other way than described above. If you want to control where the call enters your
|
|
|
|
|
; dialplan, which context, you want to define a peer with the hostname of the provider's
|
|
|
|
|
; server. If the provider has multiple servers to place calls to your system, you need
|
|
|
|
|
; a peer for each server.
|
|
|
|
|
;
|
2009-01-13 21:18:13 +00:00
|
|
|
; Beginning with Asterisk version 1.6.2, the "user" portion of the register line may
|
|
|
|
|
; contain a port number. Since the logical separator between a host and port number is a
|
|
|
|
|
; ':' character, and this character is already used to separate between the optional "secret"
|
|
|
|
|
; and "authuser" portions of the line, there is a bit of a hoop to jump through if you wish
|
|
|
|
|
; to use a port here. That is, you must explicitly provide a "secret" and "authuser" even if
|
|
|
|
|
; they are blank. See the third example below for an illustration.
|
|
|
|
|
;
|
|
|
|
|
;
|
2004-03-25 20:52:57 +00:00
|
|
|
; Examples:
|
2004-04-03 22:59:12 +00:00
|
|
|
;
|
2009-05-28 14:39:21 +00:00
|
|
|
;register => 1234:password@mysipprovider.com
|
2004-03-25 20:52:57 +00:00
|
|
|
;
|
2004-05-24 15:09:34 +00:00
|
|
|
; This will pass incoming calls to the 's' extension
|
2004-03-25 20:52:57 +00:00
|
|
|
;
|
|
|
|
|
;
|
2004-05-24 15:09:34 +00:00
|
|
|
;register => 2345:password@sip_proxy/1234
|
2004-03-25 20:52:57 +00:00
|
|
|
;
|
2005-10-04 22:51:59 +00:00
|
|
|
; Register 2345 at sip provider 'sip_proxy'. Calls from this provider
|
|
|
|
|
; connect to local extension 1234 in extensions.conf, default context,
|
|
|
|
|
; unless you configure a [sip_proxy] section below, and configure a
|
|
|
|
|
; context.
|
|
|
|
|
; Tip 1: Avoid assigning hostname to a sip.conf section like [provider.com]
|
2008-11-03 18:02:14 +00:00
|
|
|
; Tip 2: Use separate inbound and outbound sections for SIP providers
|
2005-10-04 22:51:59 +00:00
|
|
|
; (instead of type=friend) if you have calls in both directions
|
2009-01-13 21:18:13 +00:00
|
|
|
;
|
|
|
|
|
;register => 3456@mydomain:5082::@mysipprovider.com
|
|
|
|
|
;
|
|
|
|
|
; Note that in this example, the optional authuser and secret portions have
|
|
|
|
|
; been left blank because we have specified a port in the user section
|
2010-01-07 09:14:57 +00:00
|
|
|
;
|
|
|
|
|
;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').
|
2009-05-28 14:32:03 +00:00
|
|
|
|
2008-08-14 14:15:50 +00:00
|
|
|
;registertimeout=20 ; retry registration calls every 20 seconds (default)
|
|
|
|
|
;registerattempts=10 ; Number of registration attempts before we give up
|
2009-05-28 14:39:21 +00:00
|
|
|
; 0 = continue forever, hammering the other server
|
|
|
|
|
; until it accepts the registration
|
|
|
|
|
; Default is 0 tries, continue forever
|
2009-09-07 12:41:08 +00:00
|
|
|
|
2008-10-09 01:40:49 +00:00
|
|
|
;----------------------------------------- OUTBOUND MWI SUBSCRIPTIONS -------------------------
|
|
|
|
|
; Asterisk can subscribe to receive the MWI from another SIP server and store it locally for retrieval
|
2010-12-17 17:29:09 +00:00
|
|
|
; by other phones. At this time, you can only subscribe using UDP as the transport.
|
2008-10-09 01:40:49 +00:00
|
|
|
; Format for the mwi register statement is:
|
2010-12-17 17:29:09 +00:00
|
|
|
; mwi => user[:secret[:authuser]]@host[:port]/mailbox
|
2008-10-09 01:40:49 +00:00
|
|
|
;
|
|
|
|
|
; Examples:
|
|
|
|
|
;mwi => 1234:password@mysipprovider.com/1234
|
2010-12-17 17:29:09 +00:00
|
|
|
;mwi => 1234:password@myportprovider.com:6969/1234
|
|
|
|
|
;mwi => 1234:password:authuser@myauthprovider.com/1234
|
|
|
|
|
;mwi => 1234:password:authuser@myauthportprovider.com:6969/1234
|
2008-10-09 01:40:49 +00:00
|
|
|
;
|
|
|
|
|
; 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
|
2005-10-04 22:51:59 +00:00
|
|
|
;----------------------------------------- NAT SUPPORT ------------------------
|
Enhance NAT support as discussed on the -dev list, i.e.:
+ extensive documentation changes both in sip.conf.sample and in the source;
+ allow "externip" and "externhost" to include a port number as well;
+ allow "bindaddr" to have a port number (making bindport unnecessary,
even though it is still present for backward compatibility);
+ introduce the new "stunaddr" parameter to specify an STUN server to
be used from the main SIP socket;
+ extend the "sip show settings" output to show all the above.
Internally:
+ change related data structures from struct in_addr to struct sockaddr_in
to store the port numbers as well;
+ reorganize ast_sip_ouraddrfor() (should also be renamed to sip_ouraddrfor()
because it is not a generic API, though it might become so if called with
a socket as an additional argument, in which case it can be moved elsewhere).
As mentioned in the documentation, media sessions still do not use STUN so the
port numbers may still be incorrect when Asterisk is behind a NAT
On passing, some of the debugging messages printing media addresses are
probably using the wrong values, but this will be checked/fixed in a
subsequent commit if needed.
Part of the following chunk in the function that handles a "sip reload" is
probably needed on previous versions as well, to avoid leaking the memory
used for the "localaddr" list:
@@ -17244,13 +17274,17 @@
/* Reset IP addresses */
memset(&bindaddr, 0, sizeof(bindaddr));
+ memset(&stunaddr, 0, sizeof(stunaddr));
+ memset(&internip, 0, sizeof(internip));
+ /* Free memory for local network address mask */
+ ---> ast_free_ha(localaddr); <-----
memset(&localaddr, 0, sizeof(localaddr));
memset(&externip, 0, sizeof(externip));
memset(&default_prefs, 0 , sizeof(default_prefs));
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@76221 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-21 01:01:10 +00:00
|
|
|
;
|
|
|
|
|
; WARNING: SIP operation behind a NAT is tricky and you really need
|
|
|
|
|
; to read and understand well the following section.
|
|
|
|
|
;
|
|
|
|
|
; When Asterisk is behind a NAT device, the "local" address (and port) that
|
|
|
|
|
; a socket is bound to has different values when seen from the inside or
|
|
|
|
|
; from the outside of the NATted network. Unfortunately this address must
|
|
|
|
|
; be communicated to the outside (e.g. in SIP and SDP messages), and in
|
|
|
|
|
; order to determine the correct value Asterisk needs to know:
|
|
|
|
|
;
|
|
|
|
|
; + whether it is talking to someone "inside" or "outside" of the NATted network.
|
|
|
|
|
; This is configured by assigning the "localnet" parameter with a list
|
|
|
|
|
; of network addresses that are considered "inside" of the NATted network.
|
|
|
|
|
; IF LOCALNET IS NOT SET, THE EXTERNAL ADDRESS WILL NOT BE SET CORRECTLY.
|
|
|
|
|
; Multiple entries are allowed, e.g. a reasonable set is the following:
|
|
|
|
|
;
|
|
|
|
|
; localnet=192.168.0.0/255.255.0.0 ; RFC 1918 addresses
|
2008-08-14 14:15:50 +00:00
|
|
|
; localnet=10.0.0.0/255.0.0.0 ; Also RFC1918
|
|
|
|
|
; localnet=172.16.0.0/12 ; Another RFC1918 with CIDR notation
|
Enhance NAT support as discussed on the -dev list, i.e.:
+ extensive documentation changes both in sip.conf.sample and in the source;
+ allow "externip" and "externhost" to include a port number as well;
+ allow "bindaddr" to have a port number (making bindport unnecessary,
even though it is still present for backward compatibility);
+ introduce the new "stunaddr" parameter to specify an STUN server to
be used from the main SIP socket;
+ extend the "sip show settings" output to show all the above.
Internally:
+ change related data structures from struct in_addr to struct sockaddr_in
to store the port numbers as well;
+ reorganize ast_sip_ouraddrfor() (should also be renamed to sip_ouraddrfor()
because it is not a generic API, though it might become so if called with
a socket as an additional argument, in which case it can be moved elsewhere).
As mentioned in the documentation, media sessions still do not use STUN so the
port numbers may still be incorrect when Asterisk is behind a NAT
On passing, some of the debugging messages printing media addresses are
probably using the wrong values, but this will be checked/fixed in a
subsequent commit if needed.
Part of the following chunk in the function that handles a "sip reload" is
probably needed on previous versions as well, to avoid leaking the memory
used for the "localaddr" list:
@@ -17244,13 +17274,17 @@
/* Reset IP addresses */
memset(&bindaddr, 0, sizeof(bindaddr));
+ memset(&stunaddr, 0, sizeof(stunaddr));
+ memset(&internip, 0, sizeof(internip));
+ /* Free memory for local network address mask */
+ ---> ast_free_ha(localaddr); <-----
memset(&localaddr, 0, sizeof(localaddr));
memset(&externip, 0, sizeof(externip));
memset(&default_prefs, 0 , sizeof(default_prefs));
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@76221 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-21 01:01:10 +00:00
|
|
|
; localnet=169.254.0.0/255.255.0.0 ; Zero conf local network
|
|
|
|
|
;
|
|
|
|
|
; + the "externally visible" address and port number to be used when talking
|
|
|
|
|
; to a host outside the NAT. This information is derived by one of the
|
|
|
|
|
; following (mutually exclusive) config file parameters:
|
|
|
|
|
;
|
2010-07-19 17:16:23 +00:00
|
|
|
; a. "externaddr = hostname[:port]" specifies a static address[:port] to
|
Enhance NAT support as discussed on the -dev list, i.e.:
+ extensive documentation changes both in sip.conf.sample and in the source;
+ allow "externip" and "externhost" to include a port number as well;
+ allow "bindaddr" to have a port number (making bindport unnecessary,
even though it is still present for backward compatibility);
+ introduce the new "stunaddr" parameter to specify an STUN server to
be used from the main SIP socket;
+ extend the "sip show settings" output to show all the above.
Internally:
+ change related data structures from struct in_addr to struct sockaddr_in
to store the port numbers as well;
+ reorganize ast_sip_ouraddrfor() (should also be renamed to sip_ouraddrfor()
because it is not a generic API, though it might become so if called with
a socket as an additional argument, in which case it can be moved elsewhere).
As mentioned in the documentation, media sessions still do not use STUN so the
port numbers may still be incorrect when Asterisk is behind a NAT
On passing, some of the debugging messages printing media addresses are
probably using the wrong values, but this will be checked/fixed in a
subsequent commit if needed.
Part of the following chunk in the function that handles a "sip reload" is
probably needed on previous versions as well, to avoid leaking the memory
used for the "localaddr" list:
@@ -17244,13 +17274,17 @@
/* Reset IP addresses */
memset(&bindaddr, 0, sizeof(bindaddr));
+ memset(&stunaddr, 0, sizeof(stunaddr));
+ memset(&internip, 0, sizeof(internip));
+ /* Free memory for local network address mask */
+ ---> ast_free_ha(localaddr); <-----
memset(&localaddr, 0, sizeof(localaddr));
memset(&externip, 0, sizeof(externip));
memset(&default_prefs, 0 , sizeof(default_prefs));
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@76221 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-21 01:01:10 +00:00
|
|
|
; be used in SIP and SDP messages.
|
|
|
|
|
; The hostname is looked up only once, when [re]loading sip.conf .
|
2010-07-08 22:08:07 +00:00
|
|
|
; 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
|
Enhance NAT support as discussed on the -dev list, i.e.:
+ extensive documentation changes both in sip.conf.sample and in the source;
+ allow "externip" and "externhost" to include a port number as well;
+ allow "bindaddr" to have a port number (making bindport unnecessary,
even though it is still present for backward compatibility);
+ introduce the new "stunaddr" parameter to specify an STUN server to
be used from the main SIP socket;
+ extend the "sip show settings" output to show all the above.
Internally:
+ change related data structures from struct in_addr to struct sockaddr_in
to store the port numbers as well;
+ reorganize ast_sip_ouraddrfor() (should also be renamed to sip_ouraddrfor()
because it is not a generic API, though it might become so if called with
a socket as an additional argument, in which case it can be moved elsewhere).
As mentioned in the documentation, media sessions still do not use STUN so the
port numbers may still be incorrect when Asterisk is behind a NAT
On passing, some of the debugging messages printing media addresses are
probably using the wrong values, but this will be checked/fixed in a
subsequent commit if needed.
Part of the following chunk in the function that handles a "sip reload" is
probably needed on previous versions as well, to avoid leaking the memory
used for the "localaddr" list:
@@ -17244,13 +17274,17 @@
/* Reset IP addresses */
memset(&bindaddr, 0, sizeof(bindaddr));
+ memset(&stunaddr, 0, sizeof(stunaddr));
+ memset(&internip, 0, sizeof(internip));
+ /* Free memory for local network address mask */
+ ---> ast_free_ha(localaddr); <-----
memset(&localaddr, 0, sizeof(localaddr));
memset(&externip, 0, sizeof(externip));
memset(&default_prefs, 0 , sizeof(default_prefs));
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@76221 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-21 01:01:10 +00:00
|
|
|
; 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:
|
|
|
|
|
;
|
2010-07-19 17:16:23 +00:00
|
|
|
; 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.
|
2009-10-06 22:39:56 +00:00
|
|
|
; externtcpport = 9900 ; The externally mapped tcp port, when Asterisk is behind a static NAT or PAT.
|
2010-07-19 17:16:23 +00:00
|
|
|
; ; externtcpport will default to the externaddr or externhost port if either one is set.
|
2009-10-06 22:39:56 +00:00
|
|
|
; 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.
|
Enhance NAT support as discussed on the -dev list, i.e.:
+ extensive documentation changes both in sip.conf.sample and in the source;
+ allow "externip" and "externhost" to include a port number as well;
+ allow "bindaddr" to have a port number (making bindport unnecessary,
even though it is still present for backward compatibility);
+ introduce the new "stunaddr" parameter to specify an STUN server to
be used from the main SIP socket;
+ extend the "sip show settings" output to show all the above.
Internally:
+ change related data structures from struct in_addr to struct sockaddr_in
to store the port numbers as well;
+ reorganize ast_sip_ouraddrfor() (should also be renamed to sip_ouraddrfor()
because it is not a generic API, though it might become so if called with
a socket as an additional argument, in which case it can be moved elsewhere).
As mentioned in the documentation, media sessions still do not use STUN so the
port numbers may still be incorrect when Asterisk is behind a NAT
On passing, some of the debugging messages printing media addresses are
probably using the wrong values, but this will be checked/fixed in a
subsequent commit if needed.
Part of the following chunk in the function that handles a "sip reload" is
probably needed on previous versions as well, to avoid leaking the memory
used for the "localaddr" list:
@@ -17244,13 +17274,17 @@
/* Reset IP addresses */
memset(&bindaddr, 0, sizeof(bindaddr));
+ memset(&stunaddr, 0, sizeof(stunaddr));
+ memset(&internip, 0, sizeof(internip));
+ /* Free memory for local network address mask */
+ ---> ast_free_ha(localaddr); <-----
memset(&localaddr, 0, sizeof(localaddr));
memset(&externip, 0, sizeof(externip));
memset(&default_prefs, 0 , sizeof(default_prefs));
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@76221 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-21 01:01:10 +00:00
|
|
|
;
|
2010-07-19 17:16:23 +00:00
|
|
|
; b. "externhost = hostname[:port]" is similar to "externaddr" except
|
Enhance NAT support as discussed on the -dev list, i.e.:
+ extensive documentation changes both in sip.conf.sample and in the source;
+ allow "externip" and "externhost" to include a port number as well;
+ allow "bindaddr" to have a port number (making bindport unnecessary,
even though it is still present for backward compatibility);
+ introduce the new "stunaddr" parameter to specify an STUN server to
be used from the main SIP socket;
+ extend the "sip show settings" output to show all the above.
Internally:
+ change related data structures from struct in_addr to struct sockaddr_in
to store the port numbers as well;
+ reorganize ast_sip_ouraddrfor() (should also be renamed to sip_ouraddrfor()
because it is not a generic API, though it might become so if called with
a socket as an additional argument, in which case it can be moved elsewhere).
As mentioned in the documentation, media sessions still do not use STUN so the
port numbers may still be incorrect when Asterisk is behind a NAT
On passing, some of the debugging messages printing media addresses are
probably using the wrong values, but this will be checked/fixed in a
subsequent commit if needed.
Part of the following chunk in the function that handles a "sip reload" is
probably needed on previous versions as well, to avoid leaking the memory
used for the "localaddr" list:
@@ -17244,13 +17274,17 @@
/* Reset IP addresses */
memset(&bindaddr, 0, sizeof(bindaddr));
+ memset(&stunaddr, 0, sizeof(stunaddr));
+ memset(&internip, 0, sizeof(internip));
+ /* Free memory for local network address mask */
+ ---> ast_free_ha(localaddr); <-----
memset(&localaddr, 0, sizeof(localaddr));
memset(&externip, 0, sizeof(externip));
memset(&default_prefs, 0 , sizeof(default_prefs));
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@76221 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-21 01:01:10 +00:00
|
|
|
; that the hostname is looked up every "externrefresh" seconds
|
|
|
|
|
; (default 10s). This can be useful when your NAT device lets you choose
|
|
|
|
|
; the port mapping, but the IP address is dynamic.
|
|
|
|
|
; Beware, you might suffer from service disruption when the name server
|
|
|
|
|
; resolution fails. Examples:
|
|
|
|
|
;
|
2008-08-14 14:15:50 +00:00
|
|
|
; externhost=foo.dyndns.net ; refreshed periodically
|
|
|
|
|
; externrefresh=180 ; change the refresh interval
|
Enhance NAT support as discussed on the -dev list, i.e.:
+ extensive documentation changes both in sip.conf.sample and in the source;
+ allow "externip" and "externhost" to include a port number as well;
+ allow "bindaddr" to have a port number (making bindport unnecessary,
even though it is still present for backward compatibility);
+ introduce the new "stunaddr" parameter to specify an STUN server to
be used from the main SIP socket;
+ extend the "sip show settings" output to show all the above.
Internally:
+ change related data structures from struct in_addr to struct sockaddr_in
to store the port numbers as well;
+ reorganize ast_sip_ouraddrfor() (should also be renamed to sip_ouraddrfor()
because it is not a generic API, though it might become so if called with
a socket as an additional argument, in which case it can be moved elsewhere).
As mentioned in the documentation, media sessions still do not use STUN so the
port numbers may still be incorrect when Asterisk is behind a NAT
On passing, some of the debugging messages printing media addresses are
probably using the wrong values, but this will be checked/fixed in a
subsequent commit if needed.
Part of the following chunk in the function that handles a "sip reload" is
probably needed on previous versions as well, to avoid leaking the memory
used for the "localaddr" list:
@@ -17244,13 +17274,17 @@
/* Reset IP addresses */
memset(&bindaddr, 0, sizeof(bindaddr));
+ memset(&stunaddr, 0, sizeof(stunaddr));
+ memset(&internip, 0, sizeof(internip));
+ /* Free memory for local network address mask */
+ ---> ast_free_ha(localaddr); <-----
memset(&localaddr, 0, sizeof(localaddr));
memset(&externip, 0, sizeof(externip));
memset(&default_prefs, 0 , sizeof(default_prefs));
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@76221 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-21 01:01:10 +00:00
|
|
|
;
|
|
|
|
|
; Note that at the moment all these mechanism work only for the SIP socket.
|
2010-08-13 22:27:20 +00:00
|
|
|
; The IP address discovered with externaddr/externhost is reused for
|
Enhance NAT support as discussed on the -dev list, i.e.:
+ extensive documentation changes both in sip.conf.sample and in the source;
+ allow "externip" and "externhost" to include a port number as well;
+ allow "bindaddr" to have a port number (making bindport unnecessary,
even though it is still present for backward compatibility);
+ introduce the new "stunaddr" parameter to specify an STUN server to
be used from the main SIP socket;
+ extend the "sip show settings" output to show all the above.
Internally:
+ change related data structures from struct in_addr to struct sockaddr_in
to store the port numbers as well;
+ reorganize ast_sip_ouraddrfor() (should also be renamed to sip_ouraddrfor()
because it is not a generic API, though it might become so if called with
a socket as an additional argument, in which case it can be moved elsewhere).
As mentioned in the documentation, media sessions still do not use STUN so the
port numbers may still be incorrect when Asterisk is behind a NAT
On passing, some of the debugging messages printing media addresses are
probably using the wrong values, but this will be checked/fixed in a
subsequent commit if needed.
Part of the following chunk in the function that handles a "sip reload" is
probably needed on previous versions as well, to avoid leaking the memory
used for the "localaddr" list:
@@ -17244,13 +17274,17 @@
/* Reset IP addresses */
memset(&bindaddr, 0, sizeof(bindaddr));
+ memset(&stunaddr, 0, sizeof(stunaddr));
+ memset(&internip, 0, sizeof(internip));
+ /* Free memory for local network address mask */
+ ---> ast_free_ha(localaddr); <-----
memset(&localaddr, 0, sizeof(localaddr));
memset(&externip, 0, sizeof(externip));
memset(&default_prefs, 0 , sizeof(default_prefs));
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@76221 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-21 01:01:10 +00:00
|
|
|
; 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
|
2010-07-19 17:16:23 +00:00
|
|
|
; the internal<->external mapping. In these cases, the "externaddr" and
|
2010-08-13 22:27:20 +00:00
|
|
|
; "externhost" might not help you configure addresses properly.
|
Enhance NAT support as discussed on the -dev list, i.e.:
+ extensive documentation changes both in sip.conf.sample and in the source;
+ allow "externip" and "externhost" to include a port number as well;
+ allow "bindaddr" to have a port number (making bindport unnecessary,
even though it is still present for backward compatibility);
+ introduce the new "stunaddr" parameter to specify an STUN server to
be used from the main SIP socket;
+ extend the "sip show settings" output to show all the above.
Internally:
+ change related data structures from struct in_addr to struct sockaddr_in
to store the port numbers as well;
+ reorganize ast_sip_ouraddrfor() (should also be renamed to sip_ouraddrfor()
because it is not a generic API, though it might become so if called with
a socket as an additional argument, in which case it can be moved elsewhere).
As mentioned in the documentation, media sessions still do not use STUN so the
port numbers may still be incorrect when Asterisk is behind a NAT
On passing, some of the debugging messages printing media addresses are
probably using the wrong values, but this will be checked/fixed in a
subsequent commit if needed.
Part of the following chunk in the function that handles a "sip reload" is
probably needed on previous versions as well, to avoid leaking the memory
used for the "localaddr" list:
@@ -17244,13 +17274,17 @@
/* Reset IP addresses */
memset(&bindaddr, 0, sizeof(bindaddr));
+ memset(&stunaddr, 0, sizeof(stunaddr));
+ memset(&internip, 0, sizeof(internip));
+ /* Free memory for local network address mask */
+ ---> ast_free_ha(localaddr); <-----
memset(&localaddr, 0, sizeof(localaddr));
memset(&externip, 0, sizeof(externip));
memset(&default_prefs, 0 , sizeof(default_prefs));
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@76221 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-21 01:01:10 +00:00
|
|
|
;
|
2010-07-19 17:16:23 +00:00
|
|
|
; NOTE 2: when using "externaddr" or "externhost", the address part is
|
2010-08-13 22:27:20 +00:00
|
|
|
; also used as the external address for media sessions. Thus, the port
|
|
|
|
|
; information in the SDP may be wrong!
|
Enhance NAT support as discussed on the -dev list, i.e.:
+ extensive documentation changes both in sip.conf.sample and in the source;
+ allow "externip" and "externhost" to include a port number as well;
+ allow "bindaddr" to have a port number (making bindport unnecessary,
even though it is still present for backward compatibility);
+ introduce the new "stunaddr" parameter to specify an STUN server to
be used from the main SIP socket;
+ extend the "sip show settings" output to show all the above.
Internally:
+ change related data structures from struct in_addr to struct sockaddr_in
to store the port numbers as well;
+ reorganize ast_sip_ouraddrfor() (should also be renamed to sip_ouraddrfor()
because it is not a generic API, though it might become so if called with
a socket as an additional argument, in which case it can be moved elsewhere).
As mentioned in the documentation, media sessions still do not use STUN so the
port numbers may still be incorrect when Asterisk is behind a NAT
On passing, some of the debugging messages printing media addresses are
probably using the wrong values, but this will be checked/fixed in a
subsequent commit if needed.
Part of the following chunk in the function that handles a "sip reload" is
probably needed on previous versions as well, to avoid leaking the memory
used for the "localaddr" list:
@@ -17244,13 +17274,17 @@
/* Reset IP addresses */
memset(&bindaddr, 0, sizeof(bindaddr));
+ memset(&stunaddr, 0, sizeof(stunaddr));
+ memset(&internip, 0, sizeof(internip));
+ /* Free memory for local network address mask */
+ ---> ast_free_ha(localaddr); <-----
memset(&localaddr, 0, sizeof(localaddr));
memset(&externip, 0, sizeof(externip));
memset(&default_prefs, 0 , sizeof(default_prefs));
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@76221 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-21 01:01:10 +00:00
|
|
|
;
|
|
|
|
|
; In addition to the above, Asterisk has an additional "nat" parameter to
|
|
|
|
|
; address NAT-related issues in incoming SIP or media sessions.
|
|
|
|
|
; In particular, depending on the 'nat= ' settings described below, Asterisk
|
|
|
|
|
; may override the address/port information specified in the SIP/SDP messages,
|
|
|
|
|
; and use the information (sender address) supplied by the network stack instead.
|
|
|
|
|
; However, this is only useful if the external traffic can reach us.
|
|
|
|
|
; The following settings are allowed (both globally and in individual sections):
|
|
|
|
|
;
|
2009-06-26 20:19:49 +00:00
|
|
|
; nat = no ; Default. Use rport if the remote side says to use it.
|
|
|
|
|
; nat = force_rport ; Force rport to always be on.
|
2010-01-15 16:28:38 +00:00
|
|
|
; 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.
|
|
|
|
|
;
|
|
|
|
|
; 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.
|
2009-10-21 15:35:09 +00:00
|
|
|
;
|
|
|
|
|
; 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
|
2010-08-13 20:05:44 +00:00
|
|
|
;
|
|
|
|
|
; 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
|
2005-01-09 18:05:41 +00:00
|
|
|
|
2006-11-16 15:12:30 +00:00
|
|
|
;----------------------------------- MEDIA HANDLING --------------------------------
|
2009-08-03 20:48:48 +00:00
|
|
|
; By default, Asterisk tries to re-invite media streams to an optimal path. If there's
|
2006-11-16 15:12:30 +00:00
|
|
|
; no reason for Asterisk to stay in the media path, the media will be redirected.
|
2009-08-03 20:48:48 +00:00
|
|
|
; 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.
|
2006-11-16 15:12:30 +00:00
|
|
|
;
|
2009-08-03 20:48:48 +00:00
|
|
|
;directmedia=yes ; Asterisk by default tries to redirect the
|
|
|
|
|
; RTP media stream to go directly from
|
2009-05-28 14:39:21 +00:00
|
|
|
; 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
|
2009-08-03 20:48:48 +00:00
|
|
|
; behind a NAT, or for some other reason want Asterisk to
|
2009-05-28 14:39:21 +00:00
|
|
|
; 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).
|
2008-08-14 14:15:50 +00:00
|
|
|
|
2009-10-28 20:11:07 +00:00
|
|
|
; 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
|
2010-01-12 18:22:45 +00:00
|
|
|
; setting. Note that direct T.38 is not supported.
|
2009-10-28 20:11:07 +00:00
|
|
|
|
2009-08-03 20:48:48 +00:00
|
|
|
;directmedia=nonat ; An additional option is to allow media path redirection
|
2009-05-28 14:39:21 +00:00
|
|
|
; (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).
|
2008-08-14 14:15:50 +00:00
|
|
|
|
2009-08-03 20:48:48 +00:00
|
|
|
;directmedia=update ; Yet a third option... use UPDATE for media path redirection,
|
2009-05-28 14:39:21 +00:00
|
|
|
; instead of INVITE. This can be combined with 'nonat', as
|
2009-08-03 20:48:48 +00:00
|
|
|
; 'directmedia=update,nonat'. It implies 'yes'.
|
2006-02-01 13:23:59 +00:00
|
|
|
|
2009-09-07 12:41:08 +00:00
|
|
|
;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.
|
|
|
|
|
|
2010-05-20 17:54:02 +00:00
|
|
|
;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.
|
|
|
|
|
|
2008-12-17 18:49:12 +00:00
|
|
|
;ignoresdpversion=yes ; By default, Asterisk will honor the session version
|
2009-05-28 14:39:21 +00:00
|
|
|
; 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.
|
2008-12-17 18:49:12 +00:00
|
|
|
|
2009-09-07 12:41:08 +00:00
|
|
|
;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
|
2011-01-17 15:06:10 +00:00
|
|
|
;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.
|
2009-09-07 12:41:08 +00:00
|
|
|
|
2006-02-01 13:23:59 +00:00
|
|
|
;----------------------------------------- REALTIME SUPPORT ------------------------
|
|
|
|
|
; For additional information on ARA, the Asterisk Realtime Architecture,
|
2011-02-02 18:59:29 +00:00
|
|
|
; please read https://wiki.asterisk.org/wiki/display/AST/Realtime+Database+Configuration
|
2006-02-01 13:23:59 +00:00
|
|
|
;
|
2008-08-14 14:15:50 +00:00
|
|
|
;rtcachefriends=yes ; Cache realtime friends by adding them to the internal list
|
2009-05-28 14:39:21 +00:00
|
|
|
; just like friends added from the config file only on a
|
|
|
|
|
; as-needed basis? (yes|no)
|
2008-08-14 14:15:50 +00:00
|
|
|
|
|
|
|
|
;rtsavesysname=yes ; Save systemname in realtime database at registration
|
2009-05-28 14:39:21 +00:00
|
|
|
; Default= no
|
2008-08-14 14:15:50 +00:00
|
|
|
|
|
|
|
|
;rtupdate=yes ; Send registry updates to database using realtime? (yes|no)
|
2009-05-28 14:39:21 +00:00
|
|
|
; 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.
|
2008-08-14 14:15:50 +00:00
|
|
|
;rtautoclear=yes ; Auto-Expire friends created on the fly on the same schedule
|
2009-05-28 14:39:21 +00:00
|
|
|
; 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.
|
2008-08-14 14:15:50 +00:00
|
|
|
|
|
|
|
|
;ignoreregexpire=yes ; Enabling this setting has two functions:
|
2009-05-28 14:39:21 +00:00
|
|
|
;
|
|
|
|
|
; 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
|
2005-02-10 20:04:42 +00:00
|
|
|
|
2006-02-01 13:23:59 +00:00
|
|
|
;----------------------------------------- SIP DOMAIN SUPPORT ------------------------
|
2005-09-26 23:14:59 +00:00
|
|
|
; Incoming INVITE and REFER messages can be matched against a list of 'allowed'
|
|
|
|
|
; domains, each of which can direct the call to a specific context if desired.
|
|
|
|
|
; By default, all domains are accepted and sent to the default context or the
|
|
|
|
|
; context associated with the user/peer placing the call.
|
2008-07-01 12:54:57 +00:00
|
|
|
; REGISTER to non-local domains will be automatically denied if a domain
|
|
|
|
|
; list is configured.
|
|
|
|
|
;
|
2005-09-26 23:14:59 +00:00
|
|
|
; Domains can be specified using:
|
|
|
|
|
; domain=<domain>[,<context>]
|
|
|
|
|
; Examples:
|
|
|
|
|
; domain=myasterisk.dom
|
|
|
|
|
; domain=customer.com,customer-context
|
|
|
|
|
;
|
|
|
|
|
; In addition, all the 'default' domains associated with a server should be
|
|
|
|
|
; added if incoming request filtering is desired.
|
|
|
|
|
; autodomain=yes
|
|
|
|
|
;
|
|
|
|
|
; To disallow requests for domains not serviced by this server:
|
|
|
|
|
; allowexternaldomains=no
|
|
|
|
|
|
2006-06-29 07:04:43 +00:00
|
|
|
;domain=mydomain.tld,mydomain-incoming
|
2009-05-28 14:39:21 +00:00
|
|
|
; Add domain and configure incoming context
|
|
|
|
|
; for external calls to this domain
|
2008-08-14 14:15:50 +00:00
|
|
|
;domain=1.2.3.4 ; Add IP address as local domain
|
2009-05-28 14:39:21 +00:00
|
|
|
; You can have several "domain" settings
|
2008-08-14 14:15:50 +00:00
|
|
|
;allowexternaldomains=no ; Disable INVITE and REFER to non-local domains
|
2009-05-28 14:39:21 +00:00
|
|
|
; Default is yes
|
2008-08-14 14:15:50 +00:00
|
|
|
;autodomain=yes ; Turn this on to have Asterisk add local host
|
2009-05-28 14:39:21 +00:00
|
|
|
; name and local IP to domain list.
|
2008-08-14 14:15:50 +00:00
|
|
|
|
|
|
|
|
; fromdomain=mydomain.tld ; When making outbound SIP INVITEs to
|
2009-05-28 14:39:21 +00:00
|
|
|
; non-peers, use your primary domain "identity"
|
|
|
|
|
; for From: headers instead of just your IP
|
|
|
|
|
; address. This is to be polite and
|
|
|
|
|
; it may be a mandatory requirement for some
|
|
|
|
|
; destinations which do not have a prior
|
|
|
|
|
; account relationship with your server.
|
2005-11-30 05:26:29 +00:00
|
|
|
|
Generic Advice of Charge.
Asterisk Generic AOC Representation
- Generic AOC encode/decode routines.
(Generic AOC must be encoded to be passed on the wire in the AST_CONTROL_AOC frame)
- AST_CONTROL_AOC frame type to represent generic encoded AOC data
- Manager events for AOC-S, AOC-D, and AOC-E messages
Asterisk App Support
- app_dial AOC-S pass-through support on call setup
- app_queue AOC-S pass-through support on call setup
AOC Unit Tests
- AOC Unit Tests for encode/decode routines
- AOC Unit Test for manager event representation.
SIP AOC Support
- Pass-through of generic AOC-D and AOC-E messages to snom phones via the
snom AOC specification.
- Creation of chan_sip page3 flags for the addition of the new
'snom_aoc_enabled' sip.conf option.
IAX AOC Support
- Natively supports AOC pass-through through the use of the new
AST_CONTROL_AOC frame type
DAHDI AOC Support
- ETSI PRI full AOC Pass-through support
- 'aoc_enable' chan_dahdi.conf option for independently enabling
pass-through of AOC-S, AOC-D, AOC-E.
- 'aoce_delayhangup' option for retrieving AOC-E on disconnect.
- DAHDI A() dial string option for requesting AOC services.
example usage:
;requests AOC-S, AOC-D, and AOC-E on call setup
exten=>1111,1,Dial(DAHDI/g1/1112/A(s,d,e))
Review: https://reviewboard.asterisk.org/r/552/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@267096 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-02 18:10:15 +00:00
|
|
|
;------------------------------ 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.
|
|
|
|
|
|
|
|
|
|
|
2006-05-31 16:56:50 +00:00
|
|
|
;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
|
|
|
|
|
; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
|
2009-05-28 14:39:21 +00:00
|
|
|
; 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.
|
2006-05-31 16:56:50 +00:00
|
|
|
|
|
|
|
|
; jbforce = no ; Forces the use of a jitterbuffer on the receive side of a SIP
|
2009-05-28 14:39:21 +00:00
|
|
|
; channel. Defaults to "no".
|
2006-05-31 16:56:50 +00:00
|
|
|
|
|
|
|
|
; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.
|
|
|
|
|
|
|
|
|
|
; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
|
2009-05-28 14:39:21 +00:00
|
|
|
; 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.
|
2006-05-31 16:56:50 +00:00
|
|
|
|
|
|
|
|
; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a SIP
|
2009-05-28 14:39:21 +00:00
|
|
|
; 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.
|
2006-05-31 16:56:50 +00:00
|
|
|
|
2010-03-02 19:08:38 +00:00
|
|
|
; 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.
|
|
|
|
|
|
2006-05-31 16:56:50 +00:00
|
|
|
; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
|
|
|
|
|
;-----------------------------------------------------------------------------------
|
|
|
|
|
|
2005-03-24 23:06:21 +00:00
|
|
|
[authentication]
|
|
|
|
|
; Global credentials for outbound calls, i.e. when a proxy challenges your
|
|
|
|
|
; Asterisk server for authentication. These credentials override
|
|
|
|
|
; any credentials in peer/register definition if realm is matched.
|
|
|
|
|
;
|
|
|
|
|
; This way, Asterisk can authenticate for outbound calls to other
|
2009-05-28 14:39:21 +00:00
|
|
|
; realms. We match realm on the proxy challenge and pick an set of
|
2005-03-24 23:06:21 +00:00
|
|
|
; credentials from this list
|
|
|
|
|
; Syntax:
|
2008-08-14 14:15:50 +00:00
|
|
|
; auth = <user>:<secret>@<realm>
|
|
|
|
|
; auth = <user>#<md5secret>@<realm>
|
2005-03-24 23:06:21 +00:00
|
|
|
; Example:
|
|
|
|
|
;auth=mark:topsecret@digium.com
|
2009-05-28 14:39:21 +00:00
|
|
|
;
|
|
|
|
|
; You may also add auth= statements to [peer] definitions
|
2005-03-24 23:06:21 +00:00
|
|
|
; Peer auth= override all other authentication settings if we match on realm
|
|
|
|
|
|
2005-10-04 22:51:59 +00:00
|
|
|
;------------------------------------------------------------------------------
|
2008-11-03 18:02:14 +00:00
|
|
|
; DEVICE CONFIGURATION
|
2009-05-28 14:39:21 +00:00
|
|
|
;
|
2008-11-03 18:02:14 +00:00
|
|
|
; The SIP channel has two types of devices, the friend and the peer.
|
|
|
|
|
; * The type=friend is a device type that accepts both incoming and outbound calls,
|
|
|
|
|
; where Asterisk match on the From: username on incoming calls.
|
|
|
|
|
; (A synonym for friend is "user"). This is a type you use for your local
|
|
|
|
|
; SIP phones.
|
|
|
|
|
; * The type=peer also handles both incoming and outbound calls. On inbound calls,
|
|
|
|
|
; Asterisk only matches on IP/port, not on names. This is mostly used for SIP
|
|
|
|
|
; trunks.
|
|
|
|
|
;
|
|
|
|
|
; For device names, we recommend using only a-z, numerics (0-9) and underscore
|
2009-05-28 14:39:21 +00:00
|
|
|
;
|
2008-11-03 18:02:14 +00:00
|
|
|
; For local phones, type=friend works most of the time
|
|
|
|
|
;
|
2009-05-28 14:39:21 +00:00
|
|
|
; If you have one-way audio, you probably have NAT problems.
|
2008-11-03 18:02:14 +00:00
|
|
|
; If Asterisk is on a public IP, and the phone is inside of a NAT device
|
|
|
|
|
; you will need to configure nat option for those phones.
|
|
|
|
|
; Also, turn on qualify=yes to keep the nat session open
|
2009-05-28 14:39:21 +00:00
|
|
|
;
|
|
|
|
|
; Configuration options available
|
|
|
|
|
; --------------------
|
2008-11-03 18:02:14 +00:00
|
|
|
; context
|
|
|
|
|
; callingpres
|
|
|
|
|
; permit
|
|
|
|
|
; deny
|
|
|
|
|
; secret
|
|
|
|
|
; md5secret
|
|
|
|
|
; remotesecret
|
|
|
|
|
; transport
|
|
|
|
|
; dtmfmode
|
2009-08-03 20:48:48 +00:00
|
|
|
; directmedia
|
2008-11-03 18:02:14 +00:00
|
|
|
; nat
|
|
|
|
|
; callgroup
|
|
|
|
|
; pickupgroup
|
|
|
|
|
; language
|
|
|
|
|
; allow
|
|
|
|
|
; disallow
|
|
|
|
|
; insecure
|
|
|
|
|
; trustrpid
|
|
|
|
|
; progressinband
|
|
|
|
|
; promiscredir
|
|
|
|
|
; useclientcode
|
|
|
|
|
; accountcode
|
|
|
|
|
; setvar
|
|
|
|
|
; callerid
|
|
|
|
|
; amaflags
|
|
|
|
|
; callcounter
|
|
|
|
|
; busylevel
|
|
|
|
|
; allowoverlap
|
|
|
|
|
; allowsubscribe
|
|
|
|
|
; allowtransfer
|
2008-12-17 18:49:12 +00:00
|
|
|
; ignoresdpversion
|
2008-11-03 18:02:14 +00:00
|
|
|
; subscribecontext
|
|
|
|
|
; template
|
|
|
|
|
; videosupport
|
|
|
|
|
; maxcallbitrate
|
|
|
|
|
; rfc2833compensate
|
|
|
|
|
; mailbox
|
|
|
|
|
; session-timers
|
|
|
|
|
; session-expires
|
|
|
|
|
; session-minse
|
|
|
|
|
; session-refresher
|
|
|
|
|
; t38pt_usertpsource
|
|
|
|
|
; regexten
|
|
|
|
|
; fromdomain
|
|
|
|
|
; fromuser
|
|
|
|
|
; host
|
|
|
|
|
; port
|
|
|
|
|
; qualify
|
|
|
|
|
; defaultip
|
|
|
|
|
; defaultuser
|
|
|
|
|
; rtptimeout
|
|
|
|
|
; rtpholdtimeout
|
|
|
|
|
; sendrpid
|
|
|
|
|
; outboundproxy
|
|
|
|
|
; rfc2833compensate
|
|
|
|
|
; callbackextension
|
|
|
|
|
; registertrying
|
|
|
|
|
; timert1
|
|
|
|
|
; timerb
|
|
|
|
|
; qualifyfreq
|
|
|
|
|
; t38pt_usertpsource
|
|
|
|
|
; contactpermit ; Limit what a host may register as (a neat trick
|
|
|
|
|
; contactdeny ; is to register at the same IP as a SIP provider,
|
|
|
|
|
; ; then call oneself, and get redirected to that
|
|
|
|
|
; ; same location).
|
2010-05-20 17:54:02 +00:00
|
|
|
; directmediapermit
|
|
|
|
|
; directmediadeny
|
2009-10-27 13:30:27 +00:00
|
|
|
; unsolicited_mailbox
|
2009-11-02 14:57:11 +00:00
|
|
|
; use_q850_reason
|
2010-07-16 10:00:58 +00:00
|
|
|
; maxforwards
|
2011-01-17 15:06:10 +00:00
|
|
|
; encryption
|
2008-01-16 21:53:10 +00:00
|
|
|
|
2004-05-24 15:09:34 +00:00
|
|
|
;[sip_proxy]
|
|
|
|
|
; For incoming calls only. Example: FWD (Free World Dialup)
|
2009-05-28 14:39:21 +00:00
|
|
|
; We match on IP address of the proxy for incoming calls
|
2005-01-09 18:05:41 +00:00
|
|
|
; since we can not match on username (caller id)
|
|
|
|
|
;type=peer
|
2004-05-24 15:09:34 +00:00
|
|
|
;context=from-fwd
|
2005-01-09 18:05:41 +00:00
|
|
|
;host=fwd.pulver.com
|
2004-05-24 15:09:34 +00:00
|
|
|
|
|
|
|
|
;[sip_proxy-out]
|
2008-08-14 14:15:50 +00:00
|
|
|
;type=peer ; we only want to call out, not be called
|
2008-11-03 15:25:35 +00:00
|
|
|
;remotesecret=guessit ; Our password to their service
|
2008-08-14 14:15:50 +00:00
|
|
|
;defaultuser=yourusername ; Authentication user for outbound proxies
|
|
|
|
|
;fromuser=yourusername ; Many SIP providers require this!
|
2009-05-28 14:39:21 +00:00
|
|
|
;fromdomain=provider.sip.domain
|
2004-05-24 15:09:34 +00:00
|
|
|
;host=box.provider.com
|
2009-05-22 21:09:45 +00:00
|
|
|
;transport=udp,tcp ; This sets the default transport type to udp for outgoing, and will
|
|
|
|
|
; ; accept both tcp and udp. The default transport type is only used for
|
|
|
|
|
; ; outbound messages until a Registration takes place. During the
|
|
|
|
|
; ; peer Registration the transport type may change to another supported
|
|
|
|
|
; ; type if the peer requests so.
|
|
|
|
|
|
2008-08-14 14:15:50 +00:00
|
|
|
;usereqphone=yes ; This provider requires ";user=phone" on URI
|
|
|
|
|
;callcounter=yes ; Enable call counter
|
|
|
|
|
;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
|
2009-05-28 14:39:21 +00:00
|
|
|
; Also used as "defaultport" in combination with "defaultip" settings
|
2004-05-24 15:09:34 +00:00
|
|
|
|
2006-10-06 15:41:12 +00:00
|
|
|
;--- sample definition for a provider
|
|
|
|
|
;[provider1]
|
|
|
|
|
;type=peer
|
|
|
|
|
;host=sip.provider1.com
|
2008-08-14 14:15:50 +00:00
|
|
|
;fromuser=4015552299 ; how your provider knows you
|
2008-11-03 15:16:33 +00:00
|
|
|
;remotesecret=youwillneverguessit ; The password we use to authenticate to them
|
|
|
|
|
;secret=gissadetdu ; The password they use to contact us
|
2008-08-14 14:15:50 +00:00
|
|
|
;callbackextension=123 ; Register with this server and require calls coming back to this extension
|
|
|
|
|
;transport=udp,tcp ; This sets the transport type to udp for outgoing, and will
|
|
|
|
|
; ; accept both tcp and udp. Default is udp. The first transport
|
|
|
|
|
; ; listed will always be used for outgoing connections.
|
2009-10-27 13:30:27 +00:00
|
|
|
;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.
|
2006-10-06 15:41:12 +00:00
|
|
|
|
2006-10-06 16:43:36 +00:00
|
|
|
;
|
|
|
|
|
; Because you might have a large number of similar sections, it is generally
|
|
|
|
|
; convenient to use templates for the common parameters, and add them
|
|
|
|
|
; the the various sections. Examples are below, and we can even leave
|
|
|
|
|
; the templates uncommented as they will not harm:
|
|
|
|
|
|
2008-08-14 14:15:50 +00:00
|
|
|
[basic-options](!) ; a template
|
2009-05-28 14:39:21 +00:00
|
|
|
dtmfmode=rfc2833
|
|
|
|
|
context=from-office
|
|
|
|
|
type=friend
|
2006-10-06 16:43:36 +00:00
|
|
|
|
2008-08-14 14:15:50 +00:00
|
|
|
[natted-phone](!,basic-options) ; another template inheriting basic-options
|
2009-05-28 14:39:21 +00:00
|
|
|
nat=yes
|
2009-08-03 20:48:48 +00:00
|
|
|
directmedia=no
|
2009-05-28 14:39:21 +00:00
|
|
|
host=dynamic
|
2006-10-06 16:43:36 +00:00
|
|
|
|
2008-08-14 14:15:50 +00:00
|
|
|
[public-phone](!,basic-options) ; another template inheriting basic-options
|
2009-05-28 14:39:21 +00:00
|
|
|
nat=no
|
2009-08-03 20:48:48 +00:00
|
|
|
directmedia=yes
|
2006-10-06 16:43:36 +00:00
|
|
|
|
2008-08-14 14:15:50 +00:00
|
|
|
[my-codecs](!) ; a template for my preferred codecs
|
2009-05-28 14:39:21 +00:00
|
|
|
disallow=all
|
|
|
|
|
allow=ilbc
|
|
|
|
|
allow=g729
|
|
|
|
|
allow=gsm
|
|
|
|
|
allow=g723
|
|
|
|
|
allow=ulaw
|
2005-01-09 18:05:41 +00:00
|
|
|
|
2008-08-14 14:15:50 +00:00
|
|
|
[ulaw-phone](!) ; and another one for ulaw-only
|
2009-05-28 14:39:21 +00:00
|
|
|
disallow=all
|
|
|
|
|
allow=ulaw
|
2006-10-06 16:43:36 +00:00
|
|
|
|
|
|
|
|
; and finally instantiate a few phones
|
|
|
|
|
;
|
|
|
|
|
; [2133](natted-phone,my-codecs)
|
2008-08-14 14:15:50 +00:00
|
|
|
; secret = peekaboo
|
2007-08-27 12:18:13 +00:00
|
|
|
; [2134](natted-phone,ulaw-phone)
|
2008-08-14 14:15:50 +00:00
|
|
|
; secret = not_very_secret
|
2007-08-27 12:18:13 +00:00
|
|
|
; [2136](public-phone,ulaw-phone)
|
2008-08-14 14:15:50 +00:00
|
|
|
; secret = not_very_secret_either
|
2006-10-06 16:43:36 +00:00
|
|
|
; ...
|
|
|
|
|
;
|
|
|
|
|
|
|
|
|
|
; Standard configurations not using templates look like this:
|
|
|
|
|
;
|
2004-05-24 15:09:34 +00:00
|
|
|
;[grandstream1]
|
2009-05-28 14:39:21 +00:00
|
|
|
;type=friend
|
2008-08-14 14:15:50 +00:00
|
|
|
;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
|
2009-05-28 14:39:21 +00:00
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; on incoming calls to Asterisk
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2008-08-14 14:15:50 +00:00
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;host=192.168.0.23 ; we have a static but private IP address
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2009-05-28 14:39:21 +00:00
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; No registration allowed
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2008-08-14 14:15:50 +00:00
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;nat=no ; there is not NAT between phone and Asterisk
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2009-08-03 20:48:48 +00:00
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;directmedia=yes ; allow RTP voice traffic to bypass Asterisk
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2008-08-14 14:15:50 +00:00
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;dtmfmode=info ; either RFC2833 or INFO for the BudgeTone
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;call-limit=1 ; permit only 1 outgoing call and 1 incoming call at a time
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2009-05-28 14:39:21 +00:00
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; from the phone to asterisk (deprecated)
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; 1 for the explicit peer, 1 for the explicit user,
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; remember that a friend equals 1 peer and 1 user in
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; memory
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; There is no combined call counter for a "friend"
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; so there's currently no way in sip.conf to limit
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; to one inbound or outbound call per phone. Use
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; the group counters in the dial plan for that.
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;
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2008-08-14 14:15:50 +00:00
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;mailbox=1234@default ; mailbox 1234 in voicemail context "default"
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;disallow=all ; need to disallow=all before we can use allow=
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;allow=ulaw ; Note: In user sections the order of codecs
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2009-05-28 14:39:21 +00:00
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; listed with allow= does NOT matter!
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2004-05-24 15:09:34 +00:00
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;allow=alaw
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2008-08-14 14:15:50 +00:00
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;allow=g723.1 ; Asterisk only supports g723.1 pass-thru!
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;allow=g729 ; Pass-thru only unless g729 license obtained
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;callingpres=allowed_passed_screen ; Set caller ID presentation
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2009-05-28 14:39:21 +00:00
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; See README.callingpres for more information
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2004-05-24 15:09:34 +00:00
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;[xlite1]
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2005-01-09 18:05:41 +00:00
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; Turn off silence suppression in X-Lite ("Transmit Silence"=YES)!
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; Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed
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2002-06-16 16:06:38 +00:00
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;type=friend
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2008-08-14 14:15:50 +00:00
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;regexten=1234 ; When they register, create extension 1234
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2004-05-24 15:09:34 +00:00
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;callerid="Jane Smith" <5678>
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2008-08-14 14:15:50 +00:00
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;host=dynamic ; This device needs to register
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;nat=yes ; X-Lite is behind a NAT router
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2009-08-03 20:48:48 +00:00
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;directmedia=no ; Typically set to NO if behind NAT
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2004-05-24 15:09:34 +00:00
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;disallow=all
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2008-08-14 14:15:50 +00:00
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;allow=gsm ; GSM consumes far less bandwidth than ulaw
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2004-05-24 15:09:34 +00:00
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;allow=ulaw
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;allow=alaw
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2008-08-14 14:15:50 +00:00
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;mailbox=1234@default,1233@default ; Subscribe to status of multiple mailboxes
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;registertrying=yes ; Send a 100 Trying when the device registers.
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2004-05-24 15:09:34 +00:00
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;[snom]
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2008-08-14 14:15:50 +00:00
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;type=friend ; Friends place calls and receive calls
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;context=from-sip ; Context for incoming calls from this user
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2004-05-24 15:09:34 +00:00
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;secret=blah
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2008-08-14 14:15:50 +00:00
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;subscribecontext=localextensions ; Only allow SUBSCRIBE for local extensions
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2009-05-28 14:39:21 +00:00
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;language=de ; Use German prompts for this user
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2008-08-14 14:15:50 +00:00
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;host=dynamic ; This peer register with us
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;dtmfmode=inband ; Choices are inband, rfc2833, or info
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;defaultip=192.168.0.59 ; IP used until peer registers
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;mailbox=1234@context,2345 ; Mailbox(-es) for message waiting indicator
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2009-05-28 14:39:21 +00:00
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;subscribemwi=yes ; Only send notifications if this phone
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; subscribes for mailbox notification
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;vmexten=voicemail ; dialplan extension to reach mailbox
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; sets the Message-Account in the MWI notify message
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; defaults to global vmexten which defaults to "asterisk"
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2004-05-24 15:09:34 +00:00
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;disallow=all
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2008-08-14 14:15:50 +00:00
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;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw!
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2004-05-24 15:09:34 +00:00
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2002-06-16 16:06:38 +00:00
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2005-01-04 05:13:32 +00:00
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;[polycom]
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2008-08-14 14:15:50 +00:00
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;type=friend ; Friends place calls and receive calls
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;context=from-sip ; Context for incoming calls from this user
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2005-01-04 05:13:32 +00:00
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;secret=blahpoly
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2008-08-14 14:15:50 +00:00
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;host=dynamic ; This peer register with us
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;dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info
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;defaultuser=polly ; Username to use in INVITE until peer registers
|
2007-12-05 13:09:47 +00:00
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;defaultip=192.168.40.123
|
2009-05-28 14:39:21 +00:00
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; Normally you do NOT need to set this parameter
|
2005-01-04 05:13:32 +00:00
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;disallow=all
|
2008-08-14 14:15:50 +00:00
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;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw!
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;progressinband=no ; Polycom phones don't work properly with "never"
|
2005-01-04 05:13:32 +00:00
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2002-06-16 16:06:38 +00:00
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;[pingtel]
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;type=friend
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;secret=blah
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;host=dynamic
|
2009-05-28 14:39:21 +00:00
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;insecure=port ; Allow matching of peer by IP address without
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; matching port number
|
2008-08-14 14:15:50 +00:00
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;insecure=invite ; Do not require authentication of incoming INVITEs
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;insecure=port,invite ; (both)
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;qualify=1000 ; Consider it down if it's 1 second to reply
|
2009-05-28 14:39:21 +00:00
|
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|
; Helps with NAT session
|
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; qualify=yes uses default value
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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
|
2006-04-28 16:42:42 +00:00
|
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;
|
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; Call group and Pickup group should be in the range from 0 to 63
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;
|
2008-08-14 14:15:50 +00:00
|
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;callgroup=1,3-4 ; We are in caller groups 1,3,4
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|
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;pickupgroup=1,3-5 ; We can do call pick-p for call group 1,3,4,5
|
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;defaultip=192.168.0.60 ; IP address to use if peer has not registered
|
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;deny=0.0.0.0/0.0.0.0 ; ACL: Control access to this account based on IP address
|
2006-06-26 16:24:43 +00:00
|
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|
;permit=192.168.0.60/255.255.255.0
|
2009-12-16 05:24:58 +00:00
|
|
|
;permit=192.168.0.60/24 ; we can also use CIDR notation for subnet masks
|
2010-07-19 14:17:16 +00:00
|
|
|
;permit=2001:db8::/32 ; IPv6 ACLs can be specified if desired. IPv6 ACLs
|
|
|
|
|
; apply only to IPv6 addresses, and IPv4 ACLs apply
|
|
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|
|
; only to IPv4 addresses.
|
2004-03-19 20:30:03 +00:00
|
|
|
|
2004-05-24 15:09:34 +00:00
|
|
|
;[cisco1]
|
2002-06-16 16:06:38 +00:00
|
|
|
;type=friend
|
|
|
|
|
;secret=blah
|
2008-08-14 14:15:50 +00:00
|
|
|
;qualify=200 ; Qualify peer is no more than 200ms away
|
|
|
|
|
;nat=yes ; This phone may be natted
|
2009-05-28 14:39:21 +00:00
|
|
|
; Send SIP and RTP to the IP address that packet is
|
|
|
|
|
; received from instead of trusting SIP headers
|
2008-08-14 14:15:50 +00:00
|
|
|
;host=dynamic ; This device registers with us
|
2009-08-03 20:48:48 +00:00
|
|
|
;directmedia=no ; Asterisk by default tries to redirect the
|
2009-05-28 14:39:21 +00:00
|
|
|
; 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).
|
2008-08-14 14:15:50 +00:00
|
|
|
;defaultip=192.168.0.4 ; IP address to use until registration
|
|
|
|
|
;defaultuser=goran ; Username to use when calling this device before registration
|
2009-05-28 14:39:21 +00:00
|
|
|
; Normally you do NOT need to set this parameter
|
2009-01-29 13:24:01 +00:00
|
|
|
;setvar=CUSTID=5678 ; Channel variable to be set for all calls from or to this device
|
2008-07-30 16:40:43 +00:00
|
|
|
;setvar=ATTENDED_TRANSFER_COMPLETE_SOUND=beep ; This channel variable will
|
2009-05-28 14:39:21 +00:00
|
|
|
; cause the given audio file to
|
|
|
|
|
; be played upon completion of
|
|
|
|
|
; an attended transfer.
|
2002-06-16 16:06:38 +00:00
|
|
|
|
2006-08-31 01:59:02 +00:00
|
|
|
;[pre14-asterisk]
|
|
|
|
|
;type=friend
|
|
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|
|
;secret=digium
|
|
|
|
|
;host=dynamic
|
2008-08-14 14:15:50 +00:00
|
|
|
;rfc2833compensate=yes ; Compensate for pre-1.4 DTMF transmission from another Asterisk machine.
|
2009-05-28 14:39:21 +00:00
|
|
|
; You must have this turned on or DTMF reception will work improperly.
|
2008-05-28 14:29:01 +00:00
|
|
|
;t38pt_usertpsource=yes ; Use the source IP address of RTP as the destination IP address for UDPTL packets
|
2009-05-28 14:39:21 +00:00
|
|
|
; 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
|
|
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|
|
; then UDPTL will flow to the remote device.
|