; Note that all configuration options except dtlsenable can be set at the general level. ;mwi_from=asterisk ; When sending MWI NOTIFY requests, use this setting in, ; the From: header as the "name" portion. The file editor is awesome. ; limits the other side's codec choice to exactly what we prefer. ;encryption_taglen=80 ; Set the auth tag length offered in the INVITE either 32/80 default 80. ;avpf=yes ; Enable inter-operability with media streams using the AVPF RTP profile. ; If left unspecified, the default is the general-. ; "externhost" might not help you configure addresses properly. ; It can be used by other phones by following the below: ; ---------------------------------------- NAT SUPPORT ------------------------, ; WARNING: SIP operation behind a NAT is tricky and you really need. ; T38FaxMaxDatagram value specified by the other endpoint, and use a configured value instead. Setting this value to a blank, ;recordofffeature=automixmon ; Default feature to use when receiving 'Record: off' header. Defaults to 'default', ;allowguest=no ; Allow or reject guest calls (default is yes), ; If your Asterisk is connected to the Internet, ; you want to check which services you offer everyone. The asterisk.conf file. The supported protocols are listed at, ; http://www.openssl.org/docs/ssl/SSL_CTX_new.html. allow a great deal of flexibility and control they can also make configuring standard scenarios like ‘trunk’ and ‘user’ more complicated than similar sip.conf scenarios. ; how SIP URI's were typically handled in 1.6.2, hence the name. sip.conf=>mysql,asterisk,sip_buddies I got the same warning from asterisk. ; context associated with the user/peer placing the call. Each SIP client that connects to Asterisk needs a definition in SIP.CONF. Calls will fail with HANGUPCAUSE=58 if. 1.8 and earlier did not, ; add the extra headers. type=friend context=INTERNO host=dynamic disallow=all allow=ulaw allow=alaw allow=g729 … When enabled, MESSAGE. ; Set to low value if you use low timeout for NAT of UDP sessions, ;qualifygap=100 ; Number of milliseconds between each group of peers being qualified, ;qualifypeers=1 ; Number of peers in a group to be qualified at the same time, ;keepalive=60 ; Interval at which keepalive packets should be sent to a peer. ; Note that at the moment all these mechanism work only for the SIP socket. You can use it to edit your own files in /etc/asterisk -bk ; SIP Session-Timers provide an end-to-end keep-alive mechanism for active SIP sessions. ; a template, [natted-phone](!,basic-options) ; another template inheriting basic-options, [public-phone](!,basic-options) ; another template inheriting basic-options, [my-codecs](!) ; jblog = no ; Enables jitterbuffer frame logging. For example, and easy example of the sip.conf file: [general] context=default port=5060 ; UDP port for Asterisk bindaddr=0.0.0.0 ; If we want to specify only an IP (if a computer has three different IPs) 0.0.0.0 means any IP Any included files will also be converted, and written out with a pjsip_ prefix, unless changed with the --prefix= ; the progress() application in the priority before the app. ][transport://]user[@domain][:secret[:authuser]]@host[:port][/extension][~expiry], ; - 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, ; If no extension is given, the 's' extension is used. Examples: ; -------------------------------------------------------------. ', and '-' not, ; in square brackets. ; Value is in milliseconds; default is 100 ms. transport=udp ; Set the default transports. GitHub Gist: instantly share code, notes, and snippets. 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. If one of the "auto" settings, ; is used in conjunction with its non-auto counterpart (nat=comedia,auto_comedia), then, ; 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. ; added if incoming request filtering is desired. (The default is port 5060 for UDP and TCP, 5061, ; The address family of the bound UDP address is used to determine how Asterisk performs, ; DNS lookups. bindport=5060 ;you can use different … ; 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, ; 1. Asterisk SIP configuration is done is sip.conf file which is located in /etc/asterisk/sip.conf. ; Multiple contexts may be specified by separating them with '&'. sip.conf; extensions.conf; Additional configuration notes for Asterisk ; … ; This is also limited to a single caller, meaning that if an. ; dtlssetup = actpass ; Whether we are willing to accept connections, connect to the other party, or both. ; *not* switch to whatever codec the callee is sending. ; 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). ; which will be empty - thus users get no ring signal. Defaults to fixed. ; Specify 'yes' to always send ringing notifications (default). Note: realtime peers will, ; probably not function across reloads in the way that you expect, if, ;rtautoclear=yes ; Auto-Expire friends created on the fly on the same schedule, ; as if it had just registered? IP PBX Configuration - Asterisk. ; If not present, defaults to 'yes'. Asterisk will always honor the 'rport' parameter if it is, ; sent. ; configuration option. Asterisk is a free and open source framework for building your own communication applications. External Address. – Bellcore-dr4 ; Specify protocol for outbound client connections. ; ---------------------------------- MEDIA HANDLING --------------------------------, ; By default, Asterisk tries to re-invite media streams to an optimal path. ; Specify 'notinuse' to only send ringing notifications for, ; extensions that are not currently in use. ; by other phones. ;tlsclientmethod=tlsv1 ; values include tlsv1, sslv3, sslv2. ; clients are on the inside of a NAT. Just as with IAX, the SIP configuration file (sip.conf) contains configuration information for SIP channels.The headings for the channel definitions are formed by a word framed in square brackets ([])—again, with the exception of the [general] section, where we define global SIP parameters.Don’t forget to use comments generously in your sip.conf file. ; contactpermit ; Limit what a host may register as (a neat trick. in SIP and SDP messages), and in. ; * session-minse - Minimum session refresh interval in seconds. The SIP channel can accept jitter, ; thus a jitterbuffer on the receive SIP side will be used only, ; jbforce = no ; Forces the use of a jitterbuffer on the receive side of a SIP. "externaddr = hostname[:port]" specifies a static address[:port] to. ; needed digits from an ambiguous dialplan match. (default: 100), ;websocket_enabled = true ; Set to false to prevent chan_sip from listening to websockets. the default is 40, so without modification, the new. Let’s start with the sip.conf file. A continución se describen los pasos a seguir para configurar los archivos sip.conf y extensions.conf para que se puedan realizar llamadas por medio del asterisk instalado en el Access Router. That is, you must explicitly provide a "secret" and "authuser" even if. To revert to 1.8- behavior, call SIPRemoveHeader with no arguments, ; before calling Transfer() to remove all additional headers from the channel. [general] allowguest=no srvlookup=no udpbindaddr=0.0.0.0 tcpenable=no canreinvite = no dtmfmode=auto [ramal-voip](!) ; AAAA records are considered. ; setting. ; this means it is necessary for the entity to register before Asterisk can call it. Discover which option is right for you. ;directmedia=update ; Yet a third option... use UPDATE for media path redirection, ; instead of INVITE. (Default is yes). The server definition for outgoing calls looks like this: In extensions.conf you’d then use a statement like this: Please note that the ${EXTEN:1} variable here extracts all but the first characters from the current extension (the current match), in this case: 9 + the following digits. ; 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. ; dtlsenable = yes ; Enable or disable DTLS-SRTP support, ; dtlsverify = yes ; Verify that provided peer certificate and fingerprint are valid, ; ; A value of 'yes' will perform both certificate and fingerprint verification, ; ; A value of 'no' will perform no certificate or fingerprint verification, ; ; A value of 'fingerprint' will perform ONLY fingerprint verification, ; ; A value of 'certificate' will perform ONLY certficiate verification, ; dtlsrekey = 60 ; Interval at which to renegotiate the TLS session and rekey the SRTP session, ; ; If this is not set or the value provided is 0 rekeying will be disabled, ; dtlsautogeneratecert = yes ; Enable ephemeral DTLS certificate generation. After following this advanced Asterisk configuration article step by step you will be able to: Just as with IAX, the SIP configuration file (sip.conf) contains configuration information for SIP channels.The headings for the channel definitions are formed by a word framed in square brackets ([])—again, with the exception of the [general] section, where we define global SIP parameters.Don’t forget to use comments generously in your sip.conf file. ; For realtime peers, when the peer is retrieved from realtime storage, ; the registration information will be used regardless of whether, ; it has expired or not; if it expires while the realtime peer, ; is still in memory (due to caching or other reasons), the, ; information will not be removed from realtime storage, ; ---------------------------------------- SIP DOMAIN SUPPORT ------------------------, ; Incoming INVITE and REFER messages can be matched against a list of 'allowed'. ; port number as well as the address). More than one regexten may be supplied if they are. 123456 or … the variable ${ALERT_INFO} can be used to create a new header called Alert-Info: which can be used to create distinctive ringing on the Cisco SIP-enabled phone devices. By default this option is disabled. ; 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. ;tos_text=af41 ; Sets TOS for RTP text packets. – Bellcore-dr1 ; you will need to configure nat option for those phones. when a proxy challenges your, ; Asterisk server for authentication. ; be reflected in this sample configuration file, as well as in the UPGRADE.txt file. ; * session-expires - Maximum session refresh interval in seconds. In sip.conf under [general] add a register definition: Format: amjad ali amjad (amjadse at yahoo dot com) 26 January 2007 00:26:45 asterisk is no doubt a nice pbx and the asteriskguru is really a guru fro nice learners. Note that directmedia ACLs are not a global, ; (There is no default setting, this is just an example), ; Use this if some of your phones are on IP addresses that, ; can not reach each other directly. To use Asterisk and OpenSER together in realtime, see Realtime Integration Of Asterisk With OpenSER. Uses the Incomplete application to, ;allowtransfer=no ; Disable all transfers (unless enabled in peers or users), ; Default is enabled. – Bellcore-Stutter ; connect to local extension 1234 in extensions.conf, default context, ; unless you configure a [sip_proxy] section below, and configure a, ; 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. ; NOTE: You cannot use the CLI to turn it off. ;compactheaders = yes ; send compact sip headers. ; to read and understand well the following section. For historical reasons, if no remotesecret is supplied for an. ;preferred_codec_only=yes ; Respond to a SIP invite with the single most preferred codec, ; rather than advertising all joint codec capabilities. Use. ; The default for Timer T1 is 500 ms or the measured run-trip time between. ; be called as long as its IP is known to Asterisk. ; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is, ; resynchronized. Common information about the channel driver is contained at the top of the configuration file, in the [general] section. ;websocket_write_timeout = 100 ; Default write timeout to set on websocket transports. ; If you have one-way audio, you probably have NAT problems. ; media streams when appropriate, even if a DTLS stream is present. El archivo sip.conf sirve para configurar todo lo relacionado con el protocolo SIP y añadir nuevos usuarios o conectar con proveedores SIP. This option can only be used in the [general] section. ; ; mailbox. ;fromuser=yourusername ; Many SIP providers require this! This is NOT an Asterisk sip.conf setting, it is used in the dialplan in conjunction with the Default Context. Similar configuration should also work for other versions of Asterisk. Precede the comment text with a semicolon; … ; variable size, actually the new jb of IAX2). Important, the Fritzbox username (Benutzername) musst only consist of number. En el mensaje INVITE que envía el servidor de Asterisk hacia el otro extremo del enlace se observa que las direcciones IP situadas en los campos Via, Contact y Connection Information en el interior del protocolo SDP, corresponden a la dirección IP pública del router, como consecuencia del uso de la variable externip en el fichero sip.conf. ;directrtpsetup=yes ; Enable the new experimental direct RTP setup. ; 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), ; register => [peer? These credentials override. Example: bindaddr=192.0.2.1, ; b) Listen on a specific IPv6 address. This. ; route-set defined by the Path headers in the REGISTER request. Connecting Two asterisk servers using SIP: We have two asterisk servers so we will start it by editing configuration files on both servers. To configure Asterisk we need to edit some configuration files in Asterisk’s directory i.e./etc/asterisk The files which we will edit are: /etc/asterisk/sip.conf If the provider has multiple servers to place calls to your system, you need, ; Beginning with Asterisk version 1.6.2, the "user" portion of the register line may, ; contain a port number. ;rtcp_mux=yes ; Enable support for RFC 5761 RTCP multiplexing which is required for, ; ---------------------------------------- REALTIME SUPPORT ------------------------. ;auth_message_requests = yes ; Enabling this option will authenticate MESSAGE requests. All product names, trademarks and registered trademarks are property of their respective owners. Before that it only supports. ;rtsavepath=yes ; If using dynamic realtime, store the path headers, ;matchexternaddrlocally = yes ; Only substitute the externaddr or externhost setting if it matches, ; your localnet setting. From a shell prompt you can type: asterisk -r -x "sip show registry" This should report your "State" as "Registered". ; t38pt_udptl = yes,fec ; Enables T.38 with FEC error correction. ; the following to any of the above strings: ; [![touser[@todomain]][![fromuser][@fromdomain]]]. ;notifycid = yes ; Control whether caller ID information is sent along with. ; Call any SIP user on the Internet, ; (Don't forget to enable DNS SRV records if you want to use this), ; 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. It includes a number of parameters relevant to Asterisk’s handling of SIP domains: [general] context = sip-in bindport = 5060 bindaddr = 192.168.20.180; sip domain settings autodomain = yes domain = smartvox.local domain = mycompany.com domain = sip1.smartvox.local,sip1-in domain = sip2.smartvox.local,sip2-in realm = … ; Multiple entries are allowed, e.g. ; and reported in milliseconds with sip show settings. Edit sip.conf in your favourite editor and add the following example configuration:; Register and get calls from Foo Provider, to our number 1-555-455-1337 register => 15554551337:password123@sip.provider.foo [fooprovider] type=friend secret=password123 username=15554551337 host=sip.provider.foo dtmfmode=rfc2833 canreinvite=no disallow=all … Specifically, one of the items mentioned is the beginnings of a multi-stream media framework. 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), ;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. ;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY, ;buggymwi=no ; Cisco SIP firmware doesn't support the MWI RFC, ; fully. ; * If set globally, not only will all peers use the Path header, but outbound REGISTER. ; like a powerloss or grandma tripping over a cable. ;tos_video=af41 ; Sets TOS for RTP video packets. In sip.conf under [general] add a register definition: Format: register => user[:secret[:authuser]]@host[:port][/extension] or register => [email protected]:[email protected] or register => [email protected]:secret:[email protected]:port/extension. You'll. ; ----------------------------- JITTER BUFFER CONFIGURATION --------------------------, ; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a, ; SIP channel. You can select "Detect Network Settings" to have the PBX detect its External and Local networks, … SIP.conf – General option in SIP.conf SIP Configuration – general. ; makes the assumption that the endpoint supports all known SIP methods. # echo > /etc/asterisk/sip.conf. ; With the current situation, you can do one of four things: ; a) Listen on a specific IPv4 address. If you use Asterisk, then the configuration required on your server is quite straightforward. Example: bindaddr=2001:db8::1, ; c) Listen on the IPv4 wildcard. 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, ; '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'. In the case of host=dynamic. If the chains. This effectively makes. ;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. ; a call in the case of a phone disappearing from the net. (and either type=peer or … It was created in 1999 by Mark Spencer, the founder of Digium, which is a privately-held company based in Huntsville, Alabama.Among other things, Digium is specialized in developing hardware for use with Asterisk. ; Note: app_voicemail mailboxes must be in the form of mailbox@context. the variable ${VXML_URL} can be used to add additional items to the To: header. Note : For our convenience I am using names for both servers … Default: rfc2833, ; 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. ;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw! For example, to set both force_rport and comedia. I have added following piece of code in my sip.conf and extensions.conf. You will have to listen quite carefully to tell that the ringing is different. ; the call directly with media peer-2-peer without re-invites. ;textsupport=no ; Support for ITU-T T.140 realtime text. ; call them) and are matched by their authorization information (authname and secret). ; NOTE: There are multiple things to consider with this setting: ; * As this influences routing of SIP requests make sure to not trust Path headers provided, ; by the user's SIP client (the proxy in front of Asterisk should remove existing user, ; * When a peer has both a path and outboundproxy set, the path will be added to Route: header.