; 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 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
+; sip debug Show all SIP messages
;
-; reload chan_sip.so Reload configuration file
-; Active SIP peers will not be reconfigured
+; module reload chan_sip.so Reload configuration file
+; Active SIP peers will not be reconfigured
;
[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
-;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
-
-;pedantic=yes ; Enable checking of tags in headers,
- ; international character conversions in URIs
- ; and multiline formatted headers for strict
- ; SIP compatibility (defaults to "no")
+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
+;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
+
+;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"
-;disallow=all ; First disallow all codecs
-;allow=ulaw ; Allow codecs in order of preference
-;allow=ilbc ; see doc/rtp-packetization for framing options
-;
+;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"
+;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
;
;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
-;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 '401 Unauthorized'
- ; instead of letting the requester know whether there was
- ; a matching user or peer for their request
-
-;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 :-(
+;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
+;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 '401 Unauthorized'
+ ; instead of letting the requester know whether there was
+ ; a matching user or peer for their request
+
+;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
; 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)
+;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
+;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) ----------------------------
; 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.
+;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 -----------------------
;
;
; Examples:
;
-;register => 1234:password@mysipprovider.com
+;register => 1234:password@mysipprovider.com
;
; This will pass incoming calls to the 's' extension
;
; 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
+;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 = 200.201.202.203 ; 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:
+;externip = 200.201.202.203 ; 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=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
; firewall's support of SIP+RTP ports. You configure Asterisk choice of RTP
; ports for incoming audio in rtp.conf
;
-;nat=no ; Global NAT settings (Affects all peers and users)
+;nat=no ; Global NAT settings (Affects all peers and users)
; yes = Always ignore info and assume NAT
; 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)
+ ; route = Assume NAT, don't send rport
+ ; (work around more UNIDEN bugs)
;----------------------------------- MEDIA HANDLING --------------------------------
; By default, Asterisk tries to re-invite the audio to an optimal path. If there's
; 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).
-
-;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'.
+;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).
+
+;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'.
-;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
+;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'.
+;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'
; allowexternaldomains=no
;domain=mydomain.tld,mydomain-incoming
- ; 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
-;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.
+ ; 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
+;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
; 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>
+; auth = <user>:<secret>@<realm>
+; auth = <user>#<md5secret>@<realm>
; Example:
;auth=mark:topsecret@digium.com
;
; User config options: Peer configuration:
; -------------------- -------------------
; context context
-; callingpres callingpres
+; callingpres callingpres
; permit permit
; deny deny
; secret secret
; 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
+; 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
; template
;host=fwd.pulver.com
;[sip_proxy-out]
-;type=peer ; we only want to call out, not be called
+;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
+;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
+;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 = 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
; 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
-;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
- ; 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!
+;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
+;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
+ ; 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
+;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
+;regexten=1234 ; When they register, create extension 1234
;callerid="Jane Smith" <5678>
-;host=dynamic ; This device needs to register
-;nat=yes ; X-Lite is behind a NAT router
-;canreinvite=no ; Typically set to NO if behind NAT
+;host=dynamic ; This device needs to register
+;nat=yes ; X-Lite is behind a NAT router
+;canreinvite=no ; Typically set to NO if behind NAT
;disallow=all
-;allow=gsm ; GSM consumes far less bandwidth than ulaw
+;allow=gsm ; GSM consumes far less bandwidth than ulaw
;allow=ulaw
;allow=alaw
-;mailbox=1234@default,1233@default ; Subscribe to status of multiple mailboxes
+;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
+;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
+;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"
+;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!
+;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
+;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
+;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"
+;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
+;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
+;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
;[cisco1]
;type=friend
;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
-;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
+;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
+;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.
+;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
projects / thirdparty / asterisk.git / commitdiff
