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Diffstat (limited to 'configs/sip.conf.sample')
| -rw-r--r-- | configs/sip.conf.sample | 642 |
1 files changed, 321 insertions, 321 deletions
diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample index 862b482d4..20467f1aa 100644 --- a/configs/sip.conf.sample +++ b/configs/sip.conf.sample @@ -3,7 +3,7 @@ ; ; SIP dial strings ;----------------------------------------------------------- -; In the dialplan (extensions.conf) you can use several +; In the dialplan (extensions.conf) you can use several ; syntaxes for dialing SIP devices. ; SIP/devicename ; SIP/username@domain (SIP uri) @@ -17,11 +17,11 @@ ; 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 +; 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] @@ -54,7 +54,7 @@ ; 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 +; 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 @@ -64,14 +64,14 @@ ; 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 +; (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 +; to "defaultuser" which is a better name, since it is used in ; combination with the "defaultip" setting. ;----------------------------------------------------------------------------- @@ -81,25 +81,25 @@ ; 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 +; You can still set limits per device in sip.conf or in a database by using ; "setvar" to set variables that can be used in the dialplan for various limits. [general] context=default ; Default context for incoming calls ;allowguest=no ; Allow or reject guest calls (default is yes) ;match_auth_username=yes ; if available, match user entry using the -; 'username' field from the authentication line -; instead of the From: field. + ; 'username' field from the authentication line + ; instead of the From: field. allowoverlap=no ; Disable overlap dialing support. (Default is yes) ;allowtransfer=no ; Disable all transfers (unless enabled in peers or users) -; Default is enabled + ; Default is enabled ;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 + ; 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 udpbindaddr=0.0.0.0 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all) -; Optionally add a port number, 192.168.1.1:5062 (default is port 5060) + ; Optionally add a port number, 192.168.1.1:5062 (default is port 5060) ; ; Note that the TCP and TLS support for chan_sip is currently considered @@ -109,20 +109,20 @@ udpbindaddr=0.0.0.0 ; IP address to bind UDP listen socket to (0.0.0 ; tcpenable=no ; Enable server for incoming TCP connections (default is no) tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces) -; Optionally add a port number, 192.168.1.1:5062 (default is port 5060) + ; Optionally add a port number, 192.168.1.1:5062 (default is port 5060) ;tlsenable=no ; Enable server for incoming TLS (secure) connections (default is no) ;tlsbindaddr=0.0.0.0 ; IP address for TLS server to bind to (0.0.0.0) binds to all interfaces) -; Optionally add a port number, 192.168.1.1:5063 (default is port 5061) -; Remember that the IP address must match the common name (hostname) in the -; certificate, so you don't want to bind a TLS socket to multiple IP addresses. + ; Optionally add a port number, 192.168.1.1:5063 (default is port 5061) + ; Remember that the IP address must match the common name (hostname) in the + ; certificate, so you don't want to bind a TLS socket to multiple IP addresses. ;tlscertfile=</path/to/certificate.pem> ; Certificate file (*.pem only) to use for TLS connections -; default is to look for "asterisk.pem" in current directory + ; default is to look for "asterisk.pem" in current directory ;tlsprivatekey=</path/to/private.pem> ; Private key file (*.pem only) for TLS connections. -; If no tlsprivatekey is specified, tlscertfile is searched for -; for both public and private key. + ; If no tlsprivatekey is specified, tlscertfile is searched for + ; for both public and private key. ;tlscafile=</path/to/certificate> ; If the server your connecting to uses a self signed certificate @@ -130,12 +130,12 @@ tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0 ; verify the authenticity of their certificate. ;tlscadir=</path/to/ca/dir> -; A directory full of CA certificates. The files must be named with -; the CA subject name hash value. -; (see man SSL_CTX_load_verify_locations for more info) +; A directory full of CA certificates. The files must be named with +; the CA subject name hash value. +; (see man SSL_CTX_load_verify_locations for more info) ;tlsdontverifyserver=[yes|no] -; If set to yes, don't verify the servers certificate when acting as +; 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. @@ -146,20 +146,20 @@ tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0 ; 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. + ; Specify protocol for outbound client connections. + ; If left unspecified, the default is sslv2. 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 + ; 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 -;pedantic=yes ; Enable checking of tags in headers, -; international character conversions in URIs -; and multiline formatted headers for strict -; SIP compatibility (defaults to "no") +;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 qos.tex or Quality of Service section of asterisk.pdf for a description of these parameters. ;tos_sip=cs3 ; Sets TOS for SIP packets. @@ -173,32 +173,32 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ;cos_text=3 ; Sets 802.1p priority for RTP text packets. ;maxexpiry=3600 ; Maximum allowed time of incoming registrations -; and subscriptions (seconds) + ; and subscriptions (seconds) ;minexpiry=60 ; Minimum length of registrations/subscriptions (default 60) ;defaultexpiry=120 ; Default length of incoming/outgoing registration ;mwiexpiry=3600 ; Expiry time for outgoing MWI subscriptions -;qualifyfreq=60 ; Qualification: How often to check for the -; host to be up in seconds -; Set to low value if you use low timeout for -; NAT of UDP sessions +;qualifyfreq=60 ; Qualification: How often to check for the + ; host to be up in seconds + ; 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 ;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. + ; 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" + ; 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" ;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. + ; 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 @@ -220,135 +220,135 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ;mohsuggest=default ; ;parkinglot=plaza ; Sets the default parking lot for call parking -; This may also be set for individual users/peers -; Parkinglots are configured in features.conf + ; 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 + ; 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 ;sendrpid = rpid ; Use the "Remote-Party-ID" header -; to send the identity of the remote party -; This is identical to sendrpid=yes + ; 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 + ; 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. + ; 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. ;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 + ; 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. + ; The default user agent string also contains the Asterisk + ; version. If you don't want to expose this, change the + ; useragent string. ;sdpsession=Asterisk PBX ; Allows you to change the SDP session name string, (s=) -; Like the useragent parameter, the default user agent string -; also contains the Asterisk version. + ; Like the useragent parameter, the default user agent string + ; also contains the Asterisk version. ;sdpowner=root ; Allows you to change the username field in the SDP owner string, (o=) -; This field MUST NOT contain spaces + ; This field MUST NOT contain spaces ;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. + ; 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 + ; 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 + ; 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] + ; 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] ;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) + ; 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". + ; 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. + ; 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 :-( + ; 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=tls://proxy.provider.domain ; same as '=proxy.provider.domain' except we try to connect with tls ; ; (could also be tcp,udp) - defining transports on the proxy line only ; ; applies for the global proxy, otherwise use the transport= option ;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. + ; 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. + ; 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. + ; register their phones. ;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 +; 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 +; context after '@'. More than one regexten may be supplied if they are ; separated by '&'. Patterns may be used in regexten. ; ;regcontext=sipregistrations ;regextenonqualify=yes ; Default "no" -; If you have qualify on and the peer becomes unreachable -; this setting will enforce inactivation of the regexten -; extension for the peer + ; If you have qualify on and the peer becomes unreachable + ; this setting will enforce inactivation of the regexten + ; extension for the peer ; ;--------------------------- SIP timers ---------------------------------------------------- -; These timers are used primarily in INVITE transactions. +; 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 + ; Defaults to 100 ms ;timert1=500 ; Default T1 timer -; Defaults to 500 ms or the measured round-trip -; time to a peer (qualify=yes). + ; 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 + ; 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 @@ -356,15 +356,15 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; 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. + ; 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) + ; 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) + ; (default is off - zero) ;--------------------------- SIP Session-Timers (RFC 4028)------------------------------------ ; SIP Session-Timers provide an end-to-end keep-alive mechanism for active SIP sessions. @@ -403,22 +403,22 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ;--------------------------- 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) + ; 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 + ; 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 +; 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. +; for a device. ; -; If you set the busylevel, we will indicate busy when we have a number of calls that +; 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 @@ -430,38 +430,38 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; ;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 + ; 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) + ; 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. + ; 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. + ; 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. + ; device too. ;----------------------------------------- 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 +; 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. +; 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. ; @@ -469,21 +469,21 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; ; Fax Detect will cause the SIP channel to jump to the 'fax' extension (if it exists) ; after T.38 is successfully negotiated. -; -; faxdetect = yes ; Default false +; +; faxdetect = 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 => [transport://]user[:secret[:authuser]]@domain[:port][/extension][~expiry] ; -; ; -; domain is either +; +; 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 +; 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 @@ -514,7 +514,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; ; Examples: ; -;register => 1234:password@mysipprovider.com +;register => 1234:password@mysipprovider.com ; ; This will pass incoming calls to the 's' extension ; @@ -536,9 +536,9 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ;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 + ; 0 = continue forever, hammering the other server + ; until it accepts the registration + ; Default is 0 tries, continue forever ;----------------------------------------- OUTBOUND MWI SUBSCRIPTIONS ------------------------- ; Asterisk can subscribe to receive the MWI from another SIP server and store it locally for retrieval ; by other phones. @@ -635,7 +635,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; nat = no ; default. Use NAT mode only according to RFC3581 (;rport) ; nat = yes ; Always ignore info and assume NAT ; nat = never ; Never attempt NAT mode or RFC3581 support -; nat = route ; route = Assume NAT, don't send rport +; nat = route ; route = Assume NAT, don't send rport ; ; (work around more UNIDEN bugs) ;----------------------------------- MEDIA HANDLING -------------------------------- @@ -645,43 +645,43 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; 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. - -; 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). + ; 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. + + ; 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). ;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. + ; 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). + ; (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'. + ; instead of INVITE. This can be combined with 'nonat', as + ; 'canreinvite=update,nonat'. It implies 'yes'. ;ignoresdpversion=yes ; By default, Asterisk will honor the session version -; number in SDP packets and will only modify the SDP -; session if the version number changes. This option will -; force asterisk to ignore the SDP session version number -; and treat all SDP data as new data. This is required -; for devices that send us non standard SDP packets -; (observed with Microsoft OCS). By default this option is -; off. + ; number in SDP packets and will only modify the SDP + ; session if the version number changes. This option will + ; force asterisk to ignore the SDP session version number + ; and treat all SDP data as new data. This is required + ; for devices that send us non standard SDP packets + ; (observed with Microsoft OCS). By default this option is + ; off. ;----------------------------------------- REALTIME SUPPORT ------------------------ ; For additional information on ARA, the Asterisk Realtime Architecture, @@ -689,38 +689,38 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; 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) + ; 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 + ; 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. + ; 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. + ; 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 + ; + ; 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' @@ -744,45 +744,45 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; allowexternaldomains=no ;domain=mydomain.tld,mydomain-incoming -; Add domain and configure incoming context -; for external calls to this domain + ; Add domain and configure incoming context + ; for external calls to this domain ;domain=1.2.3.4 ; Add IP address as local domain -; You can have several "domain" settings + ; You can have several "domain" settings ;allowexternaldomains=no ; Disable INVITE and REFER to non-local domains -; Default is yes + ; Default is yes ;autodomain=yes ; Turn this on to have Asterisk add local host -; name and local IP to domain list. + ; 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. + ; 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. + ; 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". + ; 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. + ; 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. + ; 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". ;----------------------------------------------------------------------------------- @@ -793,20 +793,20 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; 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 +; 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:topsecret@digium.com -; -; You may also add auth= statements to [peer] definitions +; +; You may also add auth= statements to [peer] definitions ; Peer auth= override all other authentication settings if we match on realm ;------------------------------------------------------------------------------ ; DEVICE CONFIGURATION -; +; ; The SIP channel has two types of devices, the friend and the peer. ; * The type=friend is a device type that accepts both incoming and outbound calls, ; where Asterisk match on the From: username on incoming calls. @@ -817,16 +817,16 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; trunks. ; ; 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 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 -; -; Configuration options available -; -------------------- +; +; Configuration options available +; -------------------- ; context ; callingpres ; permit @@ -895,7 +895,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ;[sip_proxy] ; For incoming calls only. Example: FWD (Free World Dialup) -; We match on IP address of the proxy for incoming calls +; 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 @@ -906,7 +906,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ;remotesecret=guessit ; Our password to their service ;defaultuser=yourusername ; Authentication user for outbound proxies ;fromuser=yourusername ; Many SIP providers require this! -;fromdomain=provider.sip.domain +;fromdomain=provider.sip.domain ;host=box.provider.com ;transport=udp,tcp ; This sets the default transport type to udp for outgoing, and will ; ; accept both tcp and udp. The default transport type is only used for @@ -919,7 +919,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ;busylevel=2 ; Signal busy at 2 or more calls ;outboundproxy=proxy.provider.domain ; send outbound signaling to this proxy, not directly to the peer ;port=80 ; The port number we want to connect to on the remote side -; Also used as "defaultport" in combination with "defaultip" settings + ; Also used as "defaultport" in combination with "defaultip" settings ;--- sample definition for a provider ;[provider1] @@ -940,30 +940,30 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; the templates uncommented as they will not harm: [basic-options](!) ; a template -dtmfmode=rfc2833 -context=from-office -type=friend + dtmfmode=rfc2833 + context=from-office + type=friend [natted-phone](!,basic-options) ; another template inheriting basic-options -nat=yes -canreinvite=no -host=dynamic + nat=yes + canreinvite=no + host=dynamic [public-phone](!,basic-options) ; another template inheriting basic-options -nat=no -canreinvite=yes + nat=no + canreinvite=yes [my-codecs](!) ; a template for my preferred codecs -disallow=all -allow=ilbc -allow=g729 -allow=gsm -allow=g723 -allow=ulaw + disallow=all + allow=ilbc + allow=g729 + allow=gsm + allow=g723 + allow=ulaw [ulaw-phone](!) ; and another one for ulaw-only -disallow=all -allow=ulaw + disallow=all + allow=ulaw ; and finally instantiate a few phones ; @@ -979,34 +979,34 @@ allow=ulaw ; Standard configurations not using templates look like this: ; ;[grandstream1] -;type=friend +;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 + ; on incoming calls to Asterisk ;host=192.168.0.23 ; we have a static but private IP address -; No registration allowed + ; No registration allowed ;nat=no ; there is not NAT between phone and Asterisk ;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 (deprecated) -; 1 for the explicit peer, 1 for the explicit user, -; remember that a friend equals 1 peer and 1 user in -; memory -; 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. -; + ; from the phone to asterisk (deprecated) + ; 1 for the explicit peer, 1 for the explicit user, + ; remember that a friend equals 1 peer and 1 user in + ; memory + ; 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! + ; 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 README.callingpres for more information + ; See README.callingpres for more information ;[xlite1] ; Turn off silence suppression in X-Lite ("Transmit Silence"=YES)! @@ -1029,16 +1029,16 @@ allow=ulaw ;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 +;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" +;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! @@ -1051,7 +1051,7 @@ allow=ulaw ;dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info ;defaultuser=polly ; Username to use in INVITE until peer registers ;defaultip=192.168.40.123 -; Normally you do NOT need to set this parameter + ; 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" @@ -1061,17 +1061,17 @@ allow=ulaw ;type=friend ;secret=blah ;host=dynamic -;insecure=port ; Allow matching of peer by IP address without -; matching port number +;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 -;qualifyfreq=60 ; Qualification: How often to check for the -; host to be up in seconds -; Set to low value if you use low timeout for -; NAT of UDP sessions + ; Helps with NAT session + ; qualify=yes uses default value +;qualifyfreq=60 ; Qualification: How often to check for the + ; host to be up in seconds + ; Set to low value if you use low timeout for + ; NAT of UDP sessions ; ; Call group and Pickup group should be in the range from 0 to 63 ; @@ -1086,30 +1086,30 @@ allow=ulaw ;secret=blah ;qualify=200 ; Qualify peer is no more than 200ms away ;nat=yes ; This phone may be natted -; Send SIP and RTP to the IP address that packet is -; received from instead of trusting SIP headers + ; Send SIP and RTP to the IP address that packet is + ; received from instead of trusting SIP headers ;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). + ; 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 ;defaultuser=goran ; Username to use when calling this device before registration -; Normally you do NOT need to set this parameter + ; Normally you do NOT need to set this parameter ;setvar=CUSTID=5678 ; Channel variable to be set for all calls from or to this device ;setvar=ATTENDED_TRANSFER_COMPLETE_SOUND=beep ; This channel variable will -; cause the given audio file to -; be played upon completion of -; an attended transfer. + ; cause the given audio file to + ; be played upon completion of + ; an attended transfer. ;[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. + ; 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. + ; 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. |