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Diffstat (limited to 'configs/sip.conf.sample')
| -rw-r--r-- | configs/sip.conf.sample | 540 |
1 files changed, 270 insertions, 270 deletions
diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample index f9e656419..862b482d4 100644 --- a/configs/sip.conf.sample +++ b/configs/sip.conf.sample @@ -88,18 +88,18 @@ 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 @@ -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") +; 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 +; 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 +; 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" +; 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 @@ -220,83 +220,83 @@ 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 +; Videosupport and maxcallbitrate is settable +; for peers and users as well ;callevents=no ; generate manager events when sip ua - ; performs events (e.g. hold) +; 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 @@ -304,18 +304,18 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; ; (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 @@ -332,9 +332,9 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; ;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. @@ -342,13 +342,13 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; 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. @@ -403,11 +403,11 @@ 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 +; the moment the channel loads this configuration ;recordhistory=yes ; Record SIP history by default - ; (see sip history / sip no history) +; (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) ---------------------------- @@ -430,30 +430,30 @@ 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 ----------------------- ; @@ -533,12 +533,12 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; ; 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 - + ;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. @@ -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". ;----------------------------------------------------------------------------------- @@ -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 ; @@ -982,31 +982,31 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ;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)! @@ -1035,10 +1035,10 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ;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 +; 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" +; 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 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ;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" @@ -1062,16 +1062,16 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ;secret=blah ;host=dynamic ;insecure=port ; Allow matching of peer by IP address without - ; matching port number +; 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 +; 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 +; 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 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ;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. |