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  • Autor Tema: Configuración de Asterisk con SIP de Orange  (Leído 642 veces)

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    Desconectado jorgeboti

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    • Operador: Orange
    Configuración de Asterisk con SIP de Orange
    « en: 05 de Enero de 2017, 18:08:06 »
    Buenas a todos,
    Tras algunas semanas intentando poner a funcionar el servicio de VoIP que me ofrece la FTTH de Orange sobre FreePBX, finalmente he conseguido que funcionara. Por ese motivo, he querido compartirlo con ustedes:

    Código: [Seleccionar]
    === Configuración del Troncal ===
    CID Saliente: nuestro número en formato +349xxxxxxxx
    Nombre: cualquiera, sin restricciones.



    User-Context: from trunk


    Register String: +349xxxxxxxx@sip.orange.es:SECRET:XXXXXXXXXXXXXX@sip.orange.es@proxy2.sip.orange.es/+349xxxxxxxx
    * Importante: Es necesario crear en el archivo host de la máquina dos entradas para resolver los siguientes FQDN:
    Código: [Seleccionar]

    Además, hay que "retocar" la configuración de asterisk para que quede de la siguiente manera:

    Código: [Seleccionar]
    Registration Default Expiry: 3600

    useragent=DSL Router/DSL Router-00.96.124

    Espero que les sirva, y cualquier duda, por aquí estoy.



    La Fibra

    Configuración de Asterisk con SIP de Orange
    « en: 05 de Enero de 2017, 18:08:06 »

    Desconectado raffio

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    Re:Configuración de Asterisk con SIP de Orange
    « Respuesta #1 en: 08 de Enero de 2017, 01:35:08 »
    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.
    ;! 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@
    ; 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.

    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=
    ;  b) Listen on a specific IPv6 address.      Example: bindaddr=2001:db8::1
    ;  c) Listen on the IPv4 wildcard.            Example: bindaddr=
    ;  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=
    ;   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 =  ; IP address to bind UDP listen socket to ( binds to all)
    ; Optionally add a port number, (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 =  ; IP address for TCP server to bind to ( binds to all interfaces)
    ; Optionally add a port number, (default is port 5060)

    ;tlsenable=no                   ; Enable server for incoming TLS (secure) connections (default is no)
    ;tlsbindaddr=            ; IP address for TLS server to bind to ( binds to all interfaces)
    ; Optionally add a port number, (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.
    ; 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.
    ;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=                        ; IPv4 address literal (default port is 5060)
    ;outboundproxy=2001:db8::1                      ; IPv6 address literal (default port is 5060)
    ;outboundproxy=               ; 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=           ; Use contactpermit and contactdeny to
    ;contactpermit=  ; 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.
    ;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.

    ;        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.

    ;        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)

    ;        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.
    ;--------------------------- 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.
    ;   Multiple entries are allowed, e.g. a reasonable set is the following:
    ;      localnet= ; RFC 1918 addresses
    ;      localnet=      ; Also RFC1918
    ;      localnet=           ; Another RFC1918 with CIDR notation
    ;      localnet= ; 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 =          ; use this address.
    ;        externaddr =     ; 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 =
    ; 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
    « Última modificación: 08 de Enero de 2017, 01:38:24 por raffio »

    Desconectado madralphw

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    Re:Configuración de Asterisk con SIP de Orange
    « Respuesta #2 en: 09 de Enero de 2017, 15:26:54 »

    Espero que les sirva, y cualquier duda, por aquí estoy.



    Está genial que verifiques que funciona, la verdad que solo me faltaba el useragent, que tenía puesto uno de jazztel (pruebas ya a la desesperada) el resto así a simple vista parece igual, tenía el prack también, pero había a veces que obtenía el forbidden (supongo porque al hacerlo por dns, obtenía 2 ip's y debe hacer round robin.
    Un duda que me asalta, en el "from trunk" habría habría que añadir el "keepalive=yes" entiendo o se mantiene por el otro, no lo tengo claro (o añadirlo en opciones generales...)
    Sobre los dns, usando los dns de resolución de orange, normalmente te ofrece 2 ip's para proxy2.sip.orange.es (aunque a veces varía), si resuelves con otros servidores obtienes la cache y no los que efectivamente están funcionando, de ahí todos los problemas.
    A ver si tengo tiempo esta semana y lo vuelvo a probar, que siempre está bien tener la alternativa (para vacaciones etc...)

    Saludos ;-)

    Desconectado jorgeboti

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    Re:Configuración de Asterisk con SIP de Orange
    « Respuesta #3 en: 10 de Enero de 2017, 16:27:07 »

    Espero que les sirva, y cualquier duda, por aquí estoy.



    Está genial que verifiques que funciona, la verdad que solo me faltaba el useragent, que tenía puesto uno de jazztel (pruebas ya a la desesperada) el resto así a simple vista parece igual, tenía el prack también, pero había a veces que obtenía el forbidden (supongo porque al hacerlo por dns, obtenía 2 ip's y debe hacer round robin.
    Un duda que me asalta, en el "from trunk" habría habría que añadir el "keepalive=yes" entiendo o se mantiene por el otro, no lo tengo claro (o añadirlo en opciones generales...)
    Sobre los dns, usando los dns de resolución de orange, normalmente te ofrece 2 ip's para proxy2.sip.orange.es (aunque a veces varía), si resuelves con otros servidores obtienes la cache y no los que efectivamente están funcionando, de ahí todos los problemas.
    A ver si tengo tiempo esta semana y lo vuelvo a probar, que siempre está bien tener la alternativa (para vacaciones etc...)

    Saludos ;-)


    Esta configuración está funcionando actualmente y no he tenido problemas desde el momento de publicar el post.
    Los datos que están en el post son para anexar en la configuración GUI de FreePBX, salvo la configuración del archivo host, el resto desde la interfaz GUI.



    Desconectado kezern

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    Re:Configuración de Asterisk con SIP de Orange
    « Respuesta #4 en: 21 de Febrero de 2017, 08:32:10 »
    Hola, habéis cambiado algo en el router?
    Sustituí el livebox por un asus y he leído que en WAN -NAT Passthrough hay que desactivar el SIP Passthrough.

    Un saludo

    Desconectado sbarajas

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    Re:Configuración de Asterisk con SIP de Orange
    « Respuesta #5 en: 24 de Febrero de 2017, 13:18:56 »
    He intentado configurar mi FreePBX, con los parametros que indicas.
    Entiendo que es indispensable cambiar el Router LiveBox, por uno neutro ¿correcto?

    La Fibra

    Re:Configuración de Asterisk con SIP de Orange
    « Respuesta #5 en: 24 de Febrero de 2017, 13:18:56 »