; ; SIP Configuration example for Asterisk ; ; Syntax for specifying a SIP device in extensions.conf is ; SIP/devicename where devicename is defined in a section below. ; ; You may also use ; SIP/username@domain to 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 ; ; Useful CLI commands to check peers/users: ; sip show peers Show all SIP peers (including friends) ; sip show users Show all SIP users (including friends) ; sip show registry Show status of hosts we register with ; ; sip debug Show all SIP messages ; ; module reload chan_sip.so Reload configuration file ; Active SIP peers will not be reconfigured ; ;------- 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 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). ; ; 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. ;----------------------------------------------------------------------------- [general] context=default ; Default context for incoming calls ;allowguest=no ; Allow or reject guest calls (default is yes) allowoverlap=no ; Disable overlap dialing support. (Default is yes) ;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 bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) ; bindport is the local UDP port that Asterisk will listen on bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; 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 "no") ; See doc/ip-tos.txt 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. ;maxexpiry=3600 ; Maximum allowed time of incoming registrations ; and subscriptions (seconds) ;minexpiry=60 ; Minimum length of registrations/subscriptions (default 60) ;defaultexpiry=120 ; Default length of incoming/outgoing registration ;t1min=100 ; Minimum roundtrip time for messages to monitored hosts ; Defaults to 100 ms ;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY ;checkmwi=10 ; Default time between mailbox checks for peers ;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. ;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. ; ;disallow=all ; First disallow all codecs ;allow=ulaw ; Allow codecs in order of preference ;allow=ilbc ; see doc/rtp-packetization for framing options ; 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 ; ;language=en ; Default language setting 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 ;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 ;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 "no". Also make sure that ; the SIP peer is configured with progressinband=never. ;useragent=Asterisk PBX ; Allows you to change the user agent 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 ; 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 the 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. ;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) ;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. ;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 :-( ;matchexterniplocally = yes ; Only substitute the externip 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. ;forwardloopdetected=no ; Attempt to forward a call locally if the ; destination replies with 482 Loop Detected ; default = yes ; ; 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 ; ; 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 ;--------------------------- 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 DEBUGGING --------------------------------------------------- ;sipdebug = yes ; Turn on SIP debugging by default, from ; the moment the channel loads this configuration ;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 limit set ; for a device. When the call limit is filled, we will indicate busy. Note that ; you need at least 2 in order to be able to do attended transfers. ; ; 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 = yes ; Control whether subscriptions already INUSE get sent ; RINGING when another call is sent (default: no) ;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. ;limitonpeers = yes ; Apply call limits on peers only. This will improve ; status notification when you are using type=friend ; Inbound calls, that really apply to the user part ; of a friend will now be added to and compared with ; the peer limit instead of applying two call limits, ; one for the peer and one for the user. ; "sip show inuse" will only show active calls on ; the peer side of a "type=friend" object if this ; setting is turned on. ;----------------------------------------- T.38 FAX PASSTHROUGH SUPPORT ----------------------- ; ; This setting is available in the [general] section as well as in device configurations. ; Setting this to yes, enables T.38 fax (UDPTL) passthrough on SIP to SIP calls, provided ; both parties have T38 support enabled in their Asterisk configuration ; This has to be enabled in the general section for all devices to work. You can then ; disable it on a per device basis. ; ; T.38 faxing only works in SIP to SIP calls, with no local or agent channel being used. ; ; t38pt_udptl = yes ; Default false ; ;----------------------------------------- OUTBOUND SIP REGISTRATIONS ------------------------ ; Asterisk can register as a SIP user agent to a SIP proxy (provider) ; Format for the register statement is: ; register => user[:secret[:authuser]]@host[:port][/extension] ; ; 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). ; ; host is either a host name defined in DNS or the name of a section defined ; below. ; ; Examples: ; ;register => 1234:firstname.lastname@example.org ; ; 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 type=peer and type=user sections for SIP providers ; (instead of type=friend) if you have calls in both directions ;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 ;----------------------------------------- NAT SUPPORT ------------------------ ; The externip, externhost and localnet settings are used if you use Asterisk ; behind a NAT device to communicate with services on the outside. ;externip = 220.127.116.11 ; Address that we're going to put in outbound SIP ; messages if we're behind a NAT ; The externip and localnet is used ; when registering and communicating with other proxies ; that we're registered with ;externhost=foo.dyndns.net ; Alternatively you can specify an ; external host, and Asterisk will ; perform DNS queries periodically. Not ; recommended for production ; environments! Use externip instead ;externrefresh=10 ; How often to refresh externhost if ; used ; You may add multiple local networks. A reasonable ; set of defaults are: ;localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local networks ;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 nat= setting is used when Asterisk is on a public IP, communicating with ; devices hidden behind a NAT device (broadband router). If you have one-way ; audio problems, you usually have problems with your NAT configuration or your ; firewall's support of SIP+RTP ports. You configure Asterisk choice of RTP ; ports for incoming audio in rtp.conf ; ;nat=yes ; Global NAT settings (Affects all peers and users) ; yes = Always ignore info and assume NAT (default) ; no = Use NAT mode only according to RFC3581 (;rport) ; never = Never attempt NAT mode or RFC3581 support ; route = Assume NAT, don't send rport ; (work around more UNIDEN bugs) ; ; 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=route or nat=yes in one section and nat=no or ; nat=never in the other, then valid users with settings differing from those in the ; general section will be discoverable. ;----------------------------------- MEDIA HANDLING -------------------------------- ; By default, Asterisk tries to re-invite the audio 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 with in the case where Asterisk is outside and have ; clients on the inside of a NAT. In that case, you want to set canreinvite=nonat ; ;canreinvite=yes ; Asterisk by default tries to redirect the ; RTP media stream (audio) to go directly from ; the caller to the callee. Some devices do not ; support this (especially if one of them is behind a NAT). ; The default setting is YES. If you have all clients ; behind a NAT, or for some other reason wants Asterisk to ; stay in the audio path, you may want to turn this off. ; In Asterisk 1.4 this setting also affect direct RTP ; at call setup (a new feature in 1.4 - setting up the ; call directly between the endpoints instead of sending ; a re-INVITE). ; Additionally this option does not disable all reINVITE operations. ; It only controls Asterisk generating reINVITEs for the specific ; purpose of setting up a direct media path. If a reINVITE is ; needed to switch a media stream to inactive (when placed on ; hold) or to T.38, it will still be done, regardless of this ; setting. ;directrtpsetup=yes ; Enable the new experimental direct RTP setup. This sets up ; the call directly with media peer-2-peer without re-invites. ; Will not work for video and cases where the callee sends ; RTP payloads and fmtp headers in the 200 OK that does not match the ; callers INVITE. This will also fail if canreinvite is enabled when ; the device is actually behind NAT. ;canreinvite=nonat ; An additional option is to allow media path redirection ; (reinvite) but only when the peer where the media is being ; sent is known to not be behind a NAT (as the RTP core can ; determine it based on the apparent IP address the media ; arrives from). ;canreinvite=update ; Yet a third option... use UPDATE for media path redirection, ; instead of INVITE. This can be combined with 'nonat', as ; 'canreinvite=update,nonat'. It implies 'yes'. ;----------------------------------------- REALTIME SUPPORT ------------------------ ; For additional information on ARA, the Asterisk Realtime Architecture, ; please read realtime.txt and extconfig.txt in the /doc directory of the ; source code. ; ;rtcachefriends=yes ; Cache realtime friends by adding them to the internal list ; just like friends added from the config file only on a ; as-needed basis? (yes|no) ;rtsavesysname=yes ; Save systemname in realtime database at registration ; Default= no ;rtupdate=yes ; Send registry updates to database using realtime? (yes|no) ; If set to yes, when a SIP UA registers successfully, the ip address, ; the origination port, the registration period, and the username of ; the UA will be set to database via realtime. ; If not present, defaults to 'yes'. Note: realtime peers will ; probably not function across reloads in the way that you expect, if ; you turn this option off. ;rtautoclear=yes ; Auto-Expire friends created on the fly on the same schedule ; as if it had just registered? (yes|no|<seconds>) ; If set to yes, when the registration expires, the friend will ; vanish from the configuration until requested again. If set ; to an integer, friends expire within this number of seconds ; instead of the registration interval. ;ignoreregexpire=yes ; Enabling this setting has two functions: ; ; For non-realtime peers, when their registration expires, the ; information will _not_ be removed from memory or the Asterisk database ; if you attempt to place a call to the peer, the existing information ; will be used in spite of it having expired ; ; For realtime peers, when the peer is retrieved from realtime storage, ; the registration information will be used regardless of whether ; it has expired or not; if it expires while the realtime peer ; is still in memory (due to caching or other reasons), the ; information will not be removed from realtime storage ;----------------------------------------- SIP DOMAIN SUPPORT ------------------------ ; Incoming INVITE and REFER messages can be matched against a list of 'allowed' ; domains, each of which can direct the call to a specific context if desired. ; By default, all domains are accepted and sent to the default context or the ; context associated with the user/peer placing the call. ; REGISTER to non-local domains will be automatically denied if a domain ; list is configured. ; ; Domains can be specified using: ; domain=<domain>[,<context>] ; Examples: ; domain=myasterisk.dom ; domain=customer.com,customer-context ; ; In addition, all the 'default' domains associated with a server should be ; added if incoming request filtering is desired. ; autodomain=yes ; ; To disallow requests for domains not serviced by this server: ; allowexternaldomains=no ;domain=mydomain.tld,mydomain-incoming ; Add domain and configure incoming context ; for external calls to this domain ;domain=18.104.22.168 ; Add IP address as local domain ; You can have several "domain" settings ;allowexternaldomains=no ; Disable INVITE and REFER to non-local domains ; Default is yes ;autodomain=yes ; Turn this on to have Asterisk add local host ; name and local IP to domain list. ; fromdomain=mydomain.tld ; When making outbound SIP INVITEs to ; non-peers, use your primary domain "identity" ; for From: headers instead of just your IP ; address. This is to be polite and ; it may be a mandatory requirement for some ; destinations which do not have a prior ; account relationship with your server. ;------------------------------ JITTER BUFFER CONFIGURATION -------------------------- ; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a ; SIP channel. Defaults to "no". An enabled jitterbuffer will ; be used only if the sending side can create and the receiving ; side can not accept jitter. The SIP channel can accept jitter, ; thus a jitterbuffer on the receive SIP side will be used only ; if it is forced and enabled. ; jbforce = no ; Forces the use of a jitterbuffer on the receive side of a SIP ; channel. Defaults to "no". ; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds. ; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is ; resynchronized. Useful to improve the quality of the voice, with ; big jumps in/broken timestamps, usually sent from exotic devices ; and programs. Defaults to 1000. ; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a SIP ; channel. Two implementations are currently available - "fixed" ; (with size always equals to jbmaxsize) and "adaptive" (with ; variable size, actually the new jb of IAX2). Defaults to fixed. ; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no". ;----------------------------------------------------------------------------------- [authentication] ; Global credentials for outbound calls, i.e. when a proxy challenges your ; Asterisk server for authentication. These credentials override ; any credentials in peer/register definition if realm is matched. ; ; This way, Asterisk can authenticate for outbound calls to other ; realms. We match realm on the proxy challenge and pick an set of ; credentials from this list ; Syntax: ; auth = <user>:<secret>@<realm> ; auth = <user>#<md5secret>@<realm> ; Example: ;auth=mark:email@example.com ; ; You may also add auth= statements to [peer] definitions ; Peer auth= override all other authentication settings if we match on realm ;------------------------------------------------------------------------------ ; Users and peers have different settings available. Friends have all settings, ; since a friend is both a peer and a user ; ; User config options: Peer configuration: ; -------------------- ------------------- ; context context ; callingpres callingpres ; permit permit ; deny deny ; secret secret ; md5secret md5secret ; dtmfmode dtmfmode ; canreinvite canreinvite ; nat nat ; callgroup callgroup ; pickupgroup pickupgroup ; language language ; allow allow ; disallow disallow ; insecure insecure ; trustrpid trustrpid ; progressinband progressinband ; promiscredir promiscredir ; useclientcode useclientcode ; accountcode accountcode ; setvar setvar ; callerid callerid ; amaflags amaflags ; call-limit call-limit ; allowoverlap allowoverlap ; allowsubscribe allowsubscribe ; allowtransfer allowtransfer ; subscribecontext subscribecontext ; videosupport videosupport ; maxcallbitrate maxcallbitrate ; rfc2833compensate mailbox ; t38pt_usertpsource username ; fromdomain ; regexten ; fromuser ; host ; port ; qualify ; defaultip ; rtptimeout ; rtpholdtimeout ; sendrpid ; outboundproxy ; rfc2833compensate ; t38pt_usertpsource ; contactpermit ; Limit what a host may register as (a neat trick ; contactdeny ; is to register at the same IP as a SIP provider, ; ; then call oneself, and get redirected to that ; ; same location). ;[sip_proxy] ; For incoming calls only. Example: FWD (Free World Dialup) ; We match on IP address of the proxy for incoming calls ; since we can not match on username (caller id) ;type=peer ;context=from-fwd ;host=fwd.pulver.com ;[sip_proxy-out] ;type=peer ; we only want to call out, not be called ;secret=guessit ;username=yourusername ; Authentication user for outbound proxies ;fromuser=yourusername ; Many SIP providers require this! ;fromdomain=provider.sip.domain ;host=box.provider.com ;usereqphone=yes ; This provider requires ";user=phone" on URI ;call-limit=5 ; permit only 5 simultaneous outgoing calls to this peer ;outboundproxy=proxy.provider.domain ; send outbound signaling to this proxy, not directly to the peer ; Call-limits will not be enforced on real-time peers, ; since they are not stored in-memory ;port=80 ; The port number we want to connect to on the remote side ; Also used as "defaultport" in combination with "defaultip" settings ;------------------------------------------------------------------------------ ; Definitions of locally connected SIP devices ; ; type = user a device that authenticates to us by "from" field to place calls ; type = peer a device we place calls to or that calls us and we match by host ; type = friend two configurations (peer+user) in one ; ; For device names, we recommend using only a-z, numerics (0-9) and underscore ; ; For local phones, type=friend works most of the time ; ; If you have one-way audio, you probably have NAT problems. ; If Asterisk is on a public IP, and the phone is inside of a NAT device ; you will need to configure nat option for those phones. ; Also, turn on qualify=yes to keep the nat session open ;[grandstream1] ;type=friend ;context=from-sip ; Where to start in the dialplan when this phone calls ;callerid=John Doe <1234> ; Full caller ID, to override the phones config ; on incoming calls to Asterisk ;host=192.168.0.23 ; we have a static but private IP address ; No registration allowed ;canreinvite=yes ; allow RTP voice traffic to bypass Asterisk ;dtmfmode=info ; either RFC2833 or INFO for the BudgeTone ;call-limit=1 ; permit only 1 outgoing call and 1 incoming call at a time ; from the phone to asterisk ; 1 for the explicit peer, 1 for the explicit user, ; remember that a friend equals 1 peer and 1 user in ; memory ; This will affect your subscriptions as well. ; There is no combined call counter for a "friend" ; so there's currently no way in sip.conf to limit ; to one inbound or outbound call per phone. Use ; the group counters in the dial plan for that. ; ;mailbox=1234@default ; mailbox 1234 in voicemail context "default" ;disallow=all ; need to disallow=all before we can use allow= ;allow=ulaw ; Note: In user sections the order of codecs ; listed with allow= does NOT matter! ;allow=alaw ;allow=g723.1 ; Asterisk only supports g723.1 pass-thru! ;allow=g729 ; Pass-thru only unless g729 license obtained ;callingpres=allowed_passed_screen ; Set caller ID presentation ; See doc/callingpres.txt for more information ;[xlite1] ; Turn off silence suppression in X-Lite ("Transmit Silence"=YES)! ; Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed ;type=friend ;regexten=1234 ; When they register, create extension 1234 ;callerid="Jane Smith" <5678> ;host=dynamic ; This device needs to register ;canreinvite=no ; Typically set to NO if behind NAT ;disallow=all ;allow=gsm ; GSM consumes far less bandwidth than ulaw ;allow=ulaw ;allow=alaw ;mailbox=1234@default,1233@default ; Subscribe to status of multiple mailboxes ;[snom] ;type=friend ; Friends place calls and receive calls ;context=from-sip ; Context for incoming calls from this user ;secret=blah ;subscribecontext=localextensions ; Only allow SUBSCRIBE for local extensions ;language=de ; Use German prompts for this user ;host=dynamic ; This peer register with us ;dtmfmode=inband ; Choices are inband, rfc2833, or info ;defaultip=192.168.0.59 ; IP used until peer registers ;mailbox=1234@context,2345 ; Mailbox(-es) for message waiting indicator ;subscribemwi=yes ; Only send notifications if this phone ; subscribes for mailbox notification ;vmexten=voicemail ; dialplan extension to reach mailbox ; sets the Message-Account in the MWI notify message ; defaults to global vmexten which defaults to "asterisk" ;disallow=all ;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw! ;[polycom] ;type=friend ; Friends place calls and receive calls ;context=from-sip ; Context for incoming calls from this user ;secret=blahpoly ;host=dynamic ; This peer register with us ;dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info ;username=polly ; Username to use in INVITE until peer registers ; Normally you do NOT need to set this parameter ;disallow=all ;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw! ;progressinband=no ; Polycom phones don't work properly with "never" ;[pingtel] ;type=friend ;secret=blah ;host=dynamic ;insecure=port ; Allow matching of peer by IP address without ; matching port number ;insecure=invite ; Do not require authentication of incoming INVITEs ;insecure=port,invite ; (both) ;qualify=1000 ; Consider it down if it's 1 second to reply ; Helps with NAT session ; qualify=yes uses default value ; ; Call group and Pickup group should be in the range from 0 to 63 ; ;callgroup=1,3-4 ; We are in caller groups 1,3,4 ;pickupgroup=1,3-5 ; We can do call pick-p for call group 1,3,4,5 ;defaultip=192.168.0.60 ; IP address to use if peer has not registered ;deny=0.0.0.0/0.0.0.0 ; ACL: Control access to this account based on IP address ;permit=192.168.0.60/255.255.255.0 ;permit=192.168.0.60/24 ; we can also use CIDR notation for subnet masks ;[cisco1] ;type=friend ;secret=blah ;qualify=200 ; Qualify peer is no more than 200ms away ;host=dynamic ; This device registers with us ;canreinvite=no ; Asterisk by default tries to redirect the ; RTP media stream (audio) to go directly from ; the caller to the callee. Some devices do not ; support this (especially if one of them is ; behind a NAT). ;defaultip=192.168.0.4 ; IP address to use until registration ;username=goran ; Username to use when calling this device before registration ; Normally you do NOT need to set this parameter ;setvar=CUSTID=5678 ; Channel variable to be set for all calls from this device ;[pre14-asterisk] ;type=friend ;secret=digium ;host=dynamic ;rfc2833compensate=yes ; Compensate for pre-1.4 DTMF transmission from another Asterisk machine. ; You must have this turned on or DTMF reception will work improperly. ;t38pt_usertpsource=yes ; Use the source IP address of RTP as the destination IP address for UDPTL packets ; if the nat option is enabled. If a single RTP packet is received Asterisk will know the ; external IP address of the remote device. If port forwarding is done at the client side ; then UDPTL will flow to the remote device.