Gracias jorgeboti,
Pero podrias ser mas concreto? No entiendo donde pones esa configuracion en la gui o dentro del archivo sip.conf.
Mi arcuivo sip.conf.
[code];!
;! Automatically generated configuration file
;! Filename: sip.conf (/etc/asterisk/sip.conf)
;! Generator: Manager
;! Creation Date: Sun Jan 8 00:57:02 2017
;!
;
; SIP Configuration example for Asterisk
;
; Note: Please read the security documentation for Asterisk in order to
; understand the risks of installing Asterisk with the sample
; configuration. If your Asterisk is installed on a public
; IP address connected to the Internet, you will want to learn
; about the various security settings BEFORE you start
; Asterisk.
;
; Especially note the following settings:
; - allowguest (default enabled)
; - permit/deny/acl - IP address filters
; - contactpermit/contactdeny/contactacl - IP address filters for registrations
; - context - Which set of services you offer various users
;
; SIP dial strings
;-----------------------------------------------------------
; In the dialplan (extensions.conf) you can use several
; syntaxes for dialing SIP devices.
; SIP/devicename
; SIP/username@domain (SIP uri)
; SIP/username[:password[:md5secret[:authname[:transport]]]]@host[:port]
; SIP/devicename/extension
; SIP/devicename/extension/IPorHost
; SIP/username@domain//IPorHost
;
;
; Devicename
; devicename is defined as a peer in a section below.
;
; username@domain
; Call any SIP user on the Internet
; (Don't forget to enable DNS SRV records if you want to use this)
;
; devicename/extension
; If you define a SIP proxy as a peer below, you may call
; SIP/proxyhostname/user or SIP/user@proxyhostname
; where the proxyhostname is defined in a section below
; This syntax also works with ATA's with FXO ports
;
; SIP/username[:password[:md5secret[:authname]]]@host[:port]
; This form allows you to specify password or md5secret and authname
; without altering any authentication data in config.
; Examples:
;
; SIP/*98@mysipproxy
; SIP/sales:topsecret::account02@domain.com:5062
; SIP/12345678::bc53f0ba8ceb1ded2b70e05c3f91de4f:myname@192.168.0.1
;
; IPorHost
; The next server for this call regardless of domain/peer
;
; All of these dial strings specify the SIP request URI.
; In addition, you can specify a specific To: header by adding an
; exclamation mark after the dial string, like
;
; SIP/sales@mysipproxy!sales@edvina.net
;
; A new feature for 1.8 allows one to specify a host or IP address to use
; when routing the call. This is typically used in tandem with func_srv if
; multiple methods of reaching the same domain exist. The host or IP address
; is specified after the third slash in the dialstring. Examples:
;
; SIP/devicename/extension/IPorHost
; SIP/username@domain//IPorHost
;
; CLI Commands
; -------------------------------------------------------------
; Useful CLI commands to check peers/users:
; sip show peers Show all SIP peers (including friends)
; sip show registry Show status of hosts we register with
;
; sip set debug on Show all SIP messages
;
; sip reload Reload configuration file
; sip show settings Show the current channel configuration
;
;------- 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
; names of devices with type=user
; The name is the text between square brackets [name]
; 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
; 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).
;
; When setting up trunks, make sure there's no risk that any From: username
; (caller ID) will match any of your device names, because then Asterisk
; might match the wrong device.
;
; 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
; to "defaultuser" which is a better name, since it is used in
; combination with the "defaultip" setting.
;-----------------------------------------------------------------------------
; ** Old configuration options **
; The "call-limit" configuation option is considered old is replaced
; by new functionality. To enable callcounters, you use the new
; "callcounter" setting (for extension states in queue and subscriptions)
; You are encouraged to use the dialplan groupcount functionality
; to enforce call limits instead of using this channel-specific method.
; You can still set limits per device in sip.conf or in a database by using
; "setvar" to set variables that can be used in the dialplan for various limits.
[general]
context = public ; Default context for incoming calls. Defaults to 'default'
;allowguest=no ; Allow or reject guest calls (default is yes)
; If your Asterisk is connected to the Internet
; and you have allowguest=yes
; you want to check which services you offer everyone
; out there, by enabling them in the default context (see below).
;match_auth_username=yes ; if available, match user entry using the
; 'username' field from the authentication line
; instead of the From: field.
allowoverlap = no ; Disable overlap dialing support. (Default is yes)
;allowoverlap=yes ; Enable RFC3578 overlap dialing support.
; Can use the Incomplete application to collect the
; needed digits from an ambiguous dialplan match.
;allowoverlap=dtmf ; Enable overlap dialing support using DTMF delivery
; methods (inband, RFC2833, SIP INFO) in the early
; media phase. Uses the Incomplete application to
; collect the needed digits.
;allowtransfer=no ; Disable all transfers (unless enabled in peers or users)
; Default is enabled. The Dial() options 't' and 'T' are not
; related as to whether SIP transfers are allowed or not.
;realm=mydomain.tld ; Realm for digest authentication
; defaults to "asterisk". If you set a system name in
; asterisk.conf, it defaults to that system name
; Realms MUST be globally unique according to RFC 3261
; Set this to your host name or domain name
;domainsasrealm=no ; Use domains list as realms
; You can serve multiple Realms specifying several
; 'domain=...' directives (see below).
; In this case Realm will be based on request 'From'/'To' header
; and should match one of domain names.
; Otherwise default 'realm=...' will be used.
;recordonfeature=automixmon ; Default feature to use when receiving 'Record: on' header
; from an INFO message. Defaults to 'automon'. Works with
; dynamic features. Feature must be usable on requesting
; channel for it to work. Setting this value to a blank
; will disable it.
;recordofffeature=automixmon ; Default feature to use when receiving 'Record: off' header
; from an INFO message. Defaults to 'automon'. Works with
; dynamic features. Feature must be usable on requesting
; channel for it to work. Setting this value to a blank
; will disable it.
; With the current situation, you can do one of four things:
; a) Listen on a specific IPv4 address. Example: bindaddr=192.0.2.1
; b) Listen on a specific IPv6 address. Example: bindaddr=2001:db8::1
; c) Listen on the IPv4 wildcard. Example: bindaddr=0.0.0.0
; d) Listen on the IPv4 and IPv6 wildcards. Example: bindaddr=::
; (You can choose independently for UDP, TCP, and TLS, by specifying different values for
; "udpbindaddr", "tcpbindaddr", and "tlsbindaddr".)
; (Note that using bindaddr=:: will show only a single IPv6 socket in netstat.
; IPv4 is supported at the same time using IPv4-mapped IPv6 addresses.)
;
; You may optionally add a port number. (The default is port 5060 for UDP and TCP, 5061
; for TLS).
; IPv4 example: bindaddr=0.0.0.0:5062
; IPv6 example: bindaddr=[::]:5062
;
; The address family of the bound UDP address is used to determine how Asterisk performs
; DNS lookups. In cases a) and c) above, only A records are considered. In case b), only
; AAAA records are considered. In case d), both A and AAAA records are considered. Note,
; however, that Asterisk ignores all records except the first one. In case d), when both A
; and AAAA records are available, either an A or AAAA record will be first, and which one
; depends on the operating system. On systems using glibc, AAAA records are given
; priority.
udpbindaddr = 0.0.0.0 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all)
; Optionally add a port number, 192.168.1.1:5062 (default is port 5060)
; When a dialog is started with another SIP endpoint, the other endpoint
; should include an Allow header telling us what SIP methods the endpoint
; implements. However, some endpoints either do not include an Allow header
; or lie about what methods they implement. In the former case, Asterisk
; makes the assumption that the endpoint supports all known SIP methods.
; If you know that your SIP endpoint does not provide support for a specific
; method, then you may provide a comma-separated list of methods that your
; endpoint does not implement in the disallowed_methods option. Note that
; if your endpoint is truthful with its Allow header, then there is no need
; to set this option. This option may be set in the general section or may
; be set per endpoint. If this option is set both in the general section and
; in a peer section, then the peer setting completely overrides the general
; setting (i.e. the result is *not* the union of the two options).
;
; Note also that while Asterisk currently will parse an Allow header to learn
; what methods an endpoint supports, the only actual use for this currently
; is for determining if Asterisk may send connected line UPDATE requests and
; MESSAGE requests. Its use may be expanded in the future.
;
; disallowed_methods = UPDATE
;
; Note that the TCP and TLS support for chan_sip is currently considered
; experimental. Since it is new, all of the related configuration options are
; 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.
;
tcpenable = no ; Enable server for incoming TCP connections (default is no)
tcpbindaddr = 0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces)
; Optionally add a port number, 192.168.1.1:5062 (default is port 5060)
;tlsenable=no ; Enable server for incoming TLS (secure) connections (default is no)
;tlsbindaddr=0.0.0.0 ; IP address for TLS server to bind to (0.0.0.0) binds to all interfaces)
; Optionally add a port number, 192.168.1.1:5063 (default is port 5061)
; Remember that the IP address must match the common name (hostname) in the
; certificate, so you don't want to bind a TLS socket to multiple IP addresses.
; For details how to construct a certificate for SIP see
;
http://tools.ietf.org/html/draft-ietf-sip-domain-certs;tcpauthtimeout = 30 ; tcpauthtimeout specifies the maximum number
; of seconds a client has to authenticate. If
; the client does not authenticate beofre this
; timeout expires, the client will be
; disconnected. (default: 30 seconds)
;tcpauthlimit = 100 ; tcpauthlimit specifies the maximum number of
; unauthenticated sessions that will be allowed
; to connect at any given time. (default: 100)
;websocket_write_timeout = 100 ; Default write timeout to set on websocket transports.
; This value may need to be adjusted for connections where
; Asterisk must write a substantial amount of data and the
; receiving clients are slow to process the received information.
; Value is in milliseconds; default is 100 ms.
transport = udp ; Set the default transports. The order determines the primary default transport.
; If tcpenable=no and the transport set is tcp, we will fallback to UDP.
srvlookup = yes ; Enable DNS SRV lookups on outbound calls
subscribecontext = default
; 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
; Specifying a port in a SIP peer definition or
; when dialing outbound calls will supress SRV
; lookups for that peer or call.
;pedantic=yes ; Enable checking of tags in headers,
; international character conversions in URIs
; and multiline formatted headers for strict
; SIP compatibility (defaults to "yes")
; See
https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for a description of these parameters.
;tos_sip=cs3 ; Sets TOS for SIP packets.
;tos_audio=ef ; Sets TOS for RTP audio packets.
;tos_video=af41 ; Sets TOS for RTP video packets.
;tos_text=af41 ; Sets TOS for RTP text packets.
;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.
;maxexpiry=3600 ; Maximum allowed time of incoming registrations (seconds)
;minexpiry=60 ; Minimum length of registrations (default 60)
;defaultexpiry=120 ; Default length of incoming/outgoing registration
;submaxexpiry=3600 ; Maximum allowed time of incoming subscriptions (seconds), default: maxexpiry
;subminexpiry=60 ; Minimum length of subscriptions, default: minexpiry
;mwiexpiry=3600 ; Expiry time for outgoing MWI subscriptions
;maxforwards=70 ; Setting for the SIP Max-Forwards: header (loop prevention)
; Default value is 70
;qualifyfreq=60 ; Qualification: How often to check for the host to be up in seconds
; and reported in milliseconds with sip show settings.
; Set to low value if you use low timeout for NAT of UDP sessions
; Default: 60
;qualifygap=100 ; Number of milliseconds between each group of peers being qualified
; Default: 100
;qualifypeers=1 ; Number of peers in a group to be qualified at the same time
; Default: 1
;keepalive=60 ; Interval at which keepalive packets should be sent to a peer
; Valid options are yes (60 seconds), no, or the number of seconds.
; Default: 0
;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY
;buggymwi=no ; Cisco SIP firmware doesn't support the MWI RFC
; fully. Enable this option to not get error messages
; when sending MWI to phones with this bug.
;mwi_from=asterisk ; When sending MWI NOTIFY requests, use this setting in
; the From: header as the "name" portion. Also fill the
; "user" portion of the URI in the From: header with this
; value if no fromuser is set
; Default: empty
;vmexten=voicemail ; dialplan extension to reach mailbox sets the
; Message-Account in the MWI notify message
; defaults to "asterisk"
; Codec negotiation
;
; When Asterisk is receiving a call, the codec will initially be set to the
; first codec in the allowed codecs defined for the user receiving the call
; that the caller also indicates that it supports. But, after the caller
; starts sending RTP, Asterisk will switch to using whatever codec the caller
; is sending.
;
; When Asterisk is placing a call, the codec used will be the first codec in
; the allowed codecs that the callee indicates that it supports. Asterisk will
; *not* switch to whatever codec the callee is sending.
;
;preferred_codec_only=yes ; Respond to a SIP invite with the single most preferred codec
; rather than advertising all joint codec capabilities. This
; limits the other side's codec choice to exactly what we prefer.
;disallow=all ; First disallow all codecs
;allow=ulaw ; Allow codecs in order of preference
;allow=ilbc ; see
https://wiki.asterisk.org/wiki/display/AST/RTP+Packetization; for framing options
;autoframing=yes ; Set packetization based on the remote endpoint's (ptime)
; preferences. Defaults to no.
;
; 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
;
;parkinglot=plaza ; Sets the default parking lot for call parking
; This may also be set for individual users/peers
; Parkinglots are configured in features.conf
;language=en ; Default language setting for all users/peers
; This may also be set for individual users/peers
;tonezone=se ; Default tonezone for all users/peers
; This may also be set for individual users/peers
;relaxdtmf=yes ; Relax dtmf handling
;trustrpid = no ; If Remote-Party-ID should be trusted
;sendrpid = yes ; If Remote-Party-ID should be sent (defaults to no)
;sendrpid = rpid ; Use the "Remote-Party-ID" header
; to send the identity of the remote party
; This is identical to sendrpid=yes
;sendrpid = pai ; Use the "P-Asserted-Identity" header
; to send the identity of the remote party
;rpid_update = no ; In certain cases, the only method by which a connected line
; 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.
;trust_id_outbound = no ; Controls whether or not we trust this peer with private identity
; information (when the remote party has callingpres=prohib or equivalent).
; no - RPID/PAI headers will not be included for private peer information
; yes - RPID/PAI headers will include the private peer information. Privacy
; requirements will be indicated in a Privacy header for sendrpid=pai
; legacy - RPID/PAI will be included for private peer information. In the
; case of sendrpid=pai, private data that would be included in them
; will be anonymized. For sendrpid=rpid, private data may be included
; but the remote party's domain will be anonymized. The way legacy
; behaves may violate RFC-3325, but it follows historic behavior.
; This option is set to 'legacy' by default
;prematuremedia=no ; Some ISDN links send empty media frames before
; the call is in ringing or progress state. The SIP
; channel will then send 183 indicating early media
; which will be empty - thus users get no ring signal.
; Setting this to "yes" will stop any media before we have
; call progress (meaning the SIP channel will not send 183 Session
; Progress for early media). Default is "yes". Also make sure that
; the SIP peer is configured with progressinband=never.
;
; In order for "noanswer" applications to work, you need to run
; the progress() application in the priority before the app.
;progressinband=never ; If we should generate in-band ringing always
; use 'never' to never use in-band signalling, even in cases
; where some buggy devices might not render it
; Valid values: yes, no, never Default: never
;useragent=Asterisk PBX ; Allows you to change the user agent string
; The default user agent string also contains the Asterisk
; version. If you don't want to expose this, change the
; useragent string.
;promiscredir = no ; If yes, allows 302 or REDIR to non-local SIP address
; Note that promiscredir when redirects are made to the
; local system will cause loops since Asterisk is incapable
; of performing a "hairpin" call.
;usereqphone = no ; If yes, ";user=phone" is added to uri that contains
; a valid phone number
;dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. Default: rfc2833
; 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
;compactheaders = yes ; send compact sip headers.
;
;videosupport=yes ; Turn on support for SIP video. You need to turn this
; 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]
;textsupport=no ; Support for ITU-T T.140 realtime text.
; The default value is "no".
;maxcallbitrate=384 ; Maximum bitrate for video calls (default 384 kb/s)
; Videosupport and maxcallbitrate is settable
; for peers and users as well
;callevents=no ; generate manager events when sip ua
; performs events (e.g. hold)
;authfailureevents=no ; generate manager "peerstatus" events when peer can't
; authenticate with Asterisk. Peerstatus will be "rejected".
;alwaysauthreject = yes ; When an incoming INVITE or REGISTER is to be rejected,
; 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.
; This option is set to "yes" by default.
;auth_options_requests = yes ; Enabling this option will authenticate OPTIONS requests just like
; INVITE requests are. By default this option is disabled.
;accept_outofcall_message = no ; Disable this option to reject all MESSAGE requests outside of a
; call. By default, this option is enabled. When enabled, MESSAGE
; requests are passed in to the dialplan.
;outofcall_message_context = messages ; Context all out of dialog msgs are sent to. When this
; option is not set, the context used during peer matching
; is used. This option can be defined at both the peer and
; global level.
;auth_message_requests = yes ; Enabling this option will authenticate MESSAGE requests.
; By default this option is enabled. However, it can be disabled
; should an application desire to not load the Asterisk server with
; doing authentication and implement end to end security in the
; message body.
;g726nonstandard = yes ; If the peer negotiates G726-32 audio, use AAL2 packing
; order instead of RFC3551 packing order (this is required
; for Sipura and Grandstream ATAs, among others). This is
; contrary to the RFC3551 specification, the peer _should_
; be negotiating AAL2-G726-32 instead :-(
;outboundproxy=proxy.provider.domain ; send outbound signaling to this proxy, not directly to the devices
;outboundproxy=proxy.provider.domain:8080 ; send outbound signaling to this proxy, not directly to the devices
;outboundproxy=proxy.provider.domain,force ; Send ALL outbound signalling to proxy, ignoring route: headers
;outboundproxy=tls://proxy.provider.domain ; same as '=proxy.provider.domain' except we try to connect with tls
;outboundproxy=192.0.2.1 ; IPv4 address literal (default port is 5060)
;outboundproxy=2001:db8::1 ; IPv6 address literal (default port is 5060)
;outboundproxy=192.168.0.2.1:5062 ; IPv4 address literal with explicit port
;outboundproxy=[2001:db8::1]:5062 ; IPv6 address literal with explicit port
; ; (could also be tcp,udp) - defining transports on the proxy line only
; ; applies for the global proxy, otherwise use the transport= option
;matchexternaddrlocally = yes ; Only substitute the externaddr or externhost setting if it matches
; your localnet setting. Unless you have some sort of strange network
; setup you will not need to enable this.
;dynamic_exclude_static = yes ; Disallow all dynamic hosts from registering
; as any IP address used for staticly defined
; hosts. This helps avoid the configuration
; error of allowing your users to register at
; the same address as a SIP provider.
;contactdeny=0.0.0.0/0.0.0.0 ; Use contactpermit and contactdeny to
;contactpermit=172.16.0.0/255.255.0.0 ; restrict at what IPs your users may
; register their phones.
;contactacl=named_acl_example ; Use named ACLs defined in acl.conf
;rtp_engine=asterisk ; RTP engine to use when communicating with the device
;
; If regcontext is specified, Asterisk will dynamically create and destroy a
; NoOp priority 1 extension for a given peer who registers or unregisters with
; us and have a "regexten=" configuration item.
; Multiple contexts may be specified by separating them with '&'. The
; 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
; context after '@'. More than one regexten may be supplied if they are
; separated by '&'. Patterns may be used in regexten.
;
;regcontext=sipregistrations
;regextenonqualify=yes ; Default "no"
; If you have qualify on and the peer becomes unreachable
; this setting will enforce inactivation of the regexten
; extension for the peer
;legacy_useroption_parsing=yes ; Default "no" ; If you have this option enabled and there are semicolons
; in the user field of a sip URI, the field be truncated
; at the first semicolon seen. This effectively makes
; semicolon a non-usable character for peer names, extensions,
; and maybe other, less tested things. This can be useful
; for improving compatability with devices that like to use
; user options for whatever reason. The behavior is similar to
; how SIP URI's were typically handled in 1.6.2, hence the name.
;send_diversion=no ; Default "yes" ; Asterisk normally sends Diversion headers with certain SIP
; invites to relay data about forwarded calls. If this option
; is disabled, Asterisk won't send Diversion headers unless
; they are added manually.
; The shrinkcallerid function removes '(', ' ', ')', non-trailing '.', and '-' not
; in square brackets. For example, the caller id value 555.5555 becomes 5555555
; when this option is enabled. Disabling this option results in no modification
; of the caller id value, which is necessary when the caller id represents something
; that must be preserved. This option can only be used in the [general] section.
; By default this option is on.
;
;shrinkcallerid=yes ; on by default
;use_q850_reason = no ; Default "no"
; Set to yes add Reason header and use Reason header if it is available.
; When the Transfer() application sends a REFER SIP message, extra headers specified in
; the dialplan by way of SIPAddHeader are sent out with that message. 1.8 and earlier did not
; add the extra headers. To revert to 1.8- behavior, call SIPRemoveHeader with no arguments
; before calling Transfer() to remove all additional headers from the channel. The setting
; below is for transitional compatibility only.
;
;refer_addheaders=yes ; on by default
;autocreatepeer=no ; Allow any UAC not explicitly defined to register
; WITHOUT AUTHENTICATION. Enabling this options poses a high
; potential security risk and should be avoided unless the
; server is behind a trusted firewall.
; If set to "yes", then peers created in this fashion
; are purged during SIP reloads.
; When set to "persist", the peers created in this fashion
; are not purged during SIP reloads.
;
;------------------------ TLS settings ------------------------------------------------------------
;tlscertfile=</path/to/certificate.pem> ; Certificate chain (*.pem format only) to use for TLS connections
; The certificates must be sorted starting with the subject's certificate
; and followed by intermediate CA certificates if applicable.
; Default is to look for "asterisk.pem" in current directory
;tlsprivatekey=</path/to/private.pem> ; Private key file (*.pem format only) for TLS connections.
; If no tlsprivatekey is specified, tlscertfile is searched for
; for both public and private key.
;tlscafile=</path/to/certificate>
; If the server your connecting to uses a self signed certificate
; you should have their certificate installed here so the code can
; verify the authenticity of their certificate.
;tlscapath=</path/to/ca/dir>
; A directory full of CA certificates. The files must be named with
; the CA subject name hash value.
; (see man SSL_CTX_load_verify_locations for more info)
;tlsdontverifyserver=[yes|no]
; If set to yes, don't verify the servers certificate when acting as
; a client. If you don't have the server's CA certificate you can
; set this and it will connect without requiring tlscafile to be set.
; Default is no.
;tlscipher=<SSL cipher string>
; A string specifying which SSL ciphers to use or not use
; A list of valid SSL cipher strings can be found at:
;
http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS;
;tlsclientmethod=tlsv1 ; values include tlsv1, sslv3, sslv2.
; Specify protocol for outbound client connections.
; If left unspecified, the default is sslv2.
;
;--------------------------- SIP timers ----------------------------------------------------
; These timers are used primarily in INVITE transactions.
; 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.
;
;t1min=100 ; Minimum roundtrip time for messages to monitored hosts
; Defaults to 100 ms
;timert1=500 ; Default T1 timer
; Defaults to 500 ms or the measured round-trip
; time to a peer (qualify=yes).
;timerb=32000 ; Call setup timer. If a provisional response is not received
; in this amount of time, the call will autocongest
; Defaults to 64*timert1
;--------------------------- 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
;
;rtptimeout=60 ; Terminate call if 60 seconds of no RTP or RTCP activity
; on the audio channel
; when we're not on hold. This is to be able to hangup
; a call in the case of a phone disappearing from the net,
; like a powerloss or grandma tripping over a cable.
;rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP or RTCP activity
; on the audio channel
; when we're on hold (must be > rtptimeout)
;rtpkeepalive=<secs> ; Send keepalives in the RTP stream to keep NAT open
; (default is off - zero)
;--------------------------- 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'.
; uac - Default to the caller initially refreshing when possible
; uas - Default to the callee initially refreshing when possible
;
; Note that, due to recommendations in RFC 4028, Asterisk will always honor the other
; endpoint's preference for who will handle refreshes. Asterisk will never override the
; preferences of the other endpoint. Doing so could result in Asterisk and the endpoint
; fighting over who sends the refreshes. This holds true for the initiation of session
; timers and subsequent re-INVITE requests whether Asterisk is the caller or callee, or
; whether Asterisk is currently the refresher or not.
;
;session-timers=originate
;session-expires=600
;session-minse=90
;session-refresher=uac
;
;--------------------------- SIP DEBUGGING ---------------------------------------------------
;sipdebug = yes ; Turn on SIP debugging by default, from
; the moment the channel loads this configuration.
; NOTE: You cannot use the CLI to turn it off. You'll
; need to edit this and reload the config.
;recordhistory=yes ; Record SIP history by default
; (see sip history / sip no history)
;dumphistory=yes ; Dump SIP history at end of SIP dialogue
; SIP history is output to the DEBUG logging channel
;--------------------------- STATUS NOTIFICATIONS (SUBSCRIPTIONS) ----------------------------
; You can subscribe to the status of extensions with a "hint" priority
; (See extensions.conf.sample for examples)
; chan_sip support two major formats for notifications: dialog-info and SIMPLE
;
; You will get more detailed reports (busy etc) if you have a call counter enabled
; for a device.
;
; If you set the busylevel, we will indicate busy when we have a number of calls that
; matches the busylevel treshold.
;
; 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.
;
; Note: Subscriptions does not work if you have a realtime dialplan and use the
; realtime switch.
;
;allowsubscribe=no ; Disable support for subscriptions. (Default is yes)
;subscribecontext = default ; Set a specific context for SUBSCRIBE requests
; Useful to limit subscriptions to local extensions
; Settable per peer/user also
;notifyringing = no ; Control whether subscriptions already INUSE get sent
; RINGING when another call is sent (default: yes)
;notifyhold = yes ; Notify subscriptions on HOLD state (default: no)
; Turning on notifyringing and notifyhold will add a lot
; more database transactions if you are using realtime.
;notifycid = yes ; Control whether caller ID information is sent along with
; dialog-info+xml notifications (supported by snom phones).
; Note that this feature will only work properly when the
; incoming call is using the same extension and context that
; is being used as the hint for the called extension. This means
; that it won't work when using subscribecontext for your sip
; user or peer (if subscribecontext is different than context).
; This is also limited to a single caller, meaning that if an
; extension is ringing because multiple calls are incoming,
; only one will be used as the source of caller ID. Specify
; 'ignore-context' to ignore the called context when looking
; for the caller's channel. The default value is 'no.' Setting
; notifycid to 'ignore-context' also causes call-pickups attempted
; via SNOM's NOTIFY mechanism to set the context for the call pickup
; to PICKUPMARK.
;callcounter = yes ; Enable call counters on devices. This can be set per
; device too.
;----------------------------------------- T.38 FAX SUPPORT ----------------------------------
;
; This setting is available in the [general] section as well as in device configurations.
; Setting this to yes enables T.38 FAX (UDPTL) on SIP calls; it defaults to off.
;
; t38pt_udptl = yes ; Enables T.38 with FEC error correction.
; t38pt_udptl = yes,fec ; Enables T.38 with FEC error correction.
; t38pt_udptl = yes,redundancy ; Enables T.38 with redundancy error correction.
; t38pt_udptl = yes,none ; Enables T.38 with no error correction.
;
; In some cases, T.38 endpoints will provide a T38FaxMaxDatagram value (during T.38 setup) that
; is based on an incorrect interpretation of the T.38 recommendation, and results in failures
; because Asterisk does not believe it can send T.38 packets of a reasonable size to that
; endpoint (Cisco media gateways are one example of this situation). In these cases, during a
; T.38 call you will see warning messages on the console/in the logs from the Asterisk UDPTL
; stack complaining about lack of buffer space to send T.38 FAX packets. If this occurs, you
; can set an override (globally, or on a per-device basis) to make Asterisk ignore the
; T38FaxMaxDatagram value specified by the other endpoint, and use a configured value instead.
; This can be done by appending 'maxdatagram=<value>' to the t38pt_udptl configuration option,
; like this:
;
; t38pt_udptl = yes,fec,maxdatagram=400 ; Enables T.38 with FEC error correction and overrides
; ; the other endpoint's provided value to assume we can
; ; send 400 byte T.38 FAX packets to it.
;
; FAX detection will cause the SIP channel to jump to the 'fax' extension (if it exists)
; based one or more events being detected. The events that can be detected are an incoming
; CNG tone or an incoming T.38 re-INVITE request.
;
; faxdetect = yes ; Default 'no', 'yes' enables both CNG and T.38 detection
; faxdetect = cng ; Enables only CNG detection
; faxdetect = t38 ; Enables only T.38 detection
;
;----------------------------------------- OUTBOUND SIP REGISTRATIONS ------------------------
; Asterisk can register as a SIP user agent to a SIP proxy (provider)
; Format for the register statement is:
; register => [peer?][transport://]user[@domain][:secret[:authuser]]@host[:port][/extension][~expiry]
;
;
;
; domain is either
; - domain in DNS
; - host name in DNS
; - the name of a peer defined below or in realtime
; The domain is where you register your username, so your SIP uri you are registering to
; is username@domain
;
; 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).
;
; A similar effect can be achieved by adding a "callbackextension" option in a peer section.
; this is equivalent to having the following line in the general section:
;
; register => username:secret@host/callbackextension
;
; 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).
;
; 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.
;
; 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.
;
;
; Examples:
;
;register => 1234:password@mysipprovider.com
;
; This will pass incoming calls to the 's' extension
;
;
;register => 2345:password@sip_proxy/1234
;
; 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]
; Tip 2: Use separate inbound and outbound sections for SIP providers
; (instead of type=friend) if you have calls in both directions
;
;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
;
;register => tls://username:xxxxxx@sip-tls-proxy.example.org
;
; The 'transport' part defaults to 'udp' but may also be 'tcp', 'tls', 'ws', or 'wss'.
; Using 'udp://' explicitly is also useful in case the username part
; contains a '/' ('user/name').
;registertimeout=20 ; retry registration calls every 20 seconds (default)
;registerattempts=10 ; Number of registration attempts before we give up
; 0 = continue forever, hammering the other server
; until it accepts the registration
; Default is 0 tries, continue forever
;register_retry_403=yes ; Treat 403 responses to registrations as if they were
; 401 responses and continue retrying according to normal
; retry rules.
;----------------------------------------- OUTBOUND MWI SUBSCRIPTIONS -------------------------
; Asterisk can subscribe to receive the MWI from another SIP server and store it locally for retrieval
; by other phones. At this time, you can only subscribe using UDP as the transport.
; Format for the mwi register statement is:
; mwi => user[:secret[:authuser]]@host[:port]/mailbox
;
; Examples:
;mwi => 1234:password@mysipprovider.com/1234
;mwi => 1234:password@myportprovider.com:6969/1234
;mwi => 1234:password:authuser@myauthprovider.com/1234
;mwi => 1234:password:authuser@myauthportprovider.com:6969/1234
;
; MWI received will be stored in the 1234 mailbox of the SIP_Remote context. It can be used by other phones by following the below:
; mailbox=1234@SIP_Remote
;----------------------------------------- NAT SUPPORT ------------------------
;
; 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
; localnet=10.0.0.0/255.0.0.0 ; Also RFC1918
; localnet=172.16.0.0/12 ; Another RFC1918 with CIDR notation
; 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:
;
; a. "externaddr = hostname[:port]" specifies a static address[:port] to
; be used in SIP and SDP messages.
; The hostname is looked up only once, when [re]loading sip.conf .
; If a port number is not present, use the port specified in the "udpbindaddr"
; (which is not guaranteed to work correctly, because a NAT box might remap the
; port number as well as the address).
; This approach can be useful if you have a NAT device where you can
; configure the mapping statically. Examples:
;
; externaddr = 12.34.56.78 ; use this address.
; externaddr = 12.34.56.78:9900 ; use this address and port.
; externaddr = mynat.my.org:12600 ; Public address of my nat box.
; externtcpport = 9900 ; The externally mapped tcp port, when Asterisk is behind a static NAT or PAT.
; ; externtcpport will default to the externaddr or externhost port if either one is set.
; externtlsport = 12600 ; The externally mapped tls port, when Asterisk is behind a static NAT or PAT.
; ; externtlsport port will default to the RFC designated port of 5061.
;
; b. "externhost = hostname[:port]" is similar to "externaddr" except
; that the hostname is looked up every "externrefresh" seconds
; (default 10s). This can be useful when your NAT device lets you choose
; the port mapping, but the IP address is dynamic.
; Beware, you might suffer from service disruption when the name server
; resolution fails. Examples:
;
; externhost=foo.dyndns.net ; refreshed periodically
; externrefresh=180 ; change the refresh interval
;
; Note that at the moment all these mechanism work only for the SIP socket.
; The IP address discovered with externaddr/externhost is reused for
; media sessions as well, but the port numbers are not remapped so you
; may still experience problems.
;
; NOTE 1: in some cases, NAT boxes will use different port numbers in
; the internal<->external mapping. In these cases, the "externaddr" and
; "externhost" might not help you configure addresses properly.
;
; NOTE 2: when using "externaddr" or "externhost", the address part is
; also used as the external address for media sessions. Thus, the port
; information in the SDP may be wrong!
;
; In addition to the above, Asterisk has an additional "nat" parameter to
; address NAT-related issues in incoming SIP or media sessions.
; 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):
;
; nat = no ; Do no special NAT handling other than RFC3581
; nat = force_rport ; Pretend there was an rport parameter even if there wasn't
; nat = comedia ; Send media to the port Asterisk received it from regardless
; ; of where the SDP says to send it.
; nat = auto_force_rport ; Set the force_rport option if Asterisk detects NAT (default)
; nat = auto_comedia ; Set the comedia option if Asterisk detects NAT
;
; The nat settings can be combined. For example, to set both force_rport and comedia
; one would set nat=force_rport,comedia. If any of the comma-separated options is 'no',
; Asterisk will ignore any other settings and set nat=no. If one of the "auto" settings
; is used in conjunction with its non-auto counterpart (nat=comedia,auto_comedia), then
; the non-auto option will be ignored.
;
; The RFC 3581-defined 'rport' parameter allows a client to request that Asterisk send
; SIP responses to it via the source IP and port from which the request originated
; instead of the address/port listed in the top-most Via header. This is useful if a
; client knows that it is behind a NAT and therefore cannot guess from what address/port
; its request will be sent. Asterisk will always honor the 'rport' parameter if it is
; sent. The force_rport setting causes Asterisk to always send responses back to the
; address/port from which it received requests; even if the other side doesn't support
; adding the 'rport' parameter.
;
; '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 address/port they tell Asterisk to send RTP packets to
; for their media streams is not the actual address/port that will be used on the nearer
; side of the NAT.
;
; IT IS IMPORTANT TO NOTE that if the nat setting in the general section differs from
; the nat setting in a peer definition, then the peer username will be discoverable
; by outside parties as Asterisk will respond to different ports for defined and
; undefined peers. For this reason it is recommended to ONLY DEFINE NAT SETTINGS IN THE
; GENERAL SECTION. Specifically, if nat=force_rport in one section and nat=no in the
; other, then valid peers with settings differing from those in the general section will
; be discoverable.
;
; In addition to these settings, Asterisk *always* uses 'symmetric RTP' mode as defined by
; RFC 4961; Asterisk will always send RTP packets from the same port number it expects
; to receive them on.
;
; The IP address used for media (audio, video, and text) in the SDP can also be overridden by using
; the media_address configuration option. This is only applicable to the general section and
; can not be set per-user or per-peer.
;
; media_address = 172.16.42.1
;
; Through the use of the res_stun_monitor module, Asterisk has the ability to detect when the
; perceived external network address has changed. When the stun_monitor is installed and
; configured, chan_sip will renew all outbound registrations when the monitor detects any sort
; of network change has occurred. By default this option is enabled, but only takes effect once
; res_stun_monitor is configured. If res_stun_monitor is enabled and you wish to not
; generate all outbound registrations on a network change, use the option below to disable
; this feature.
;
; subscribe_network_change_event = yes ; on by default
;
; ICE/STUN/TURN usage can be enabled globally or on a per-peer basis using the icesupport
; configuration option. When set to yes ICE support is enabled. When set to no it is disabled.
; It is disabled by default.
;
; icesupport = yes
;----------------------------------- MEDIA HANDLING --------------------------------
; By default, Asterisk tries to re-invite media streams to an optimal path. If there's
; no reason for Asterisk to stay in the media path, the media will be redirected.
; This does not really work 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.
;
;directmedia=yes ; Asterisk by default tries to redirect the
; RTP media stream to go directly from
; the caller to the callee. Some devices do not
; support this (especially if one of them is behind a NAT).
; The default setting is YES. If you have all cli
Tema: Configuración de Asterisk con SIP de Orange (Leído 7293 veces)
