[asterisk-commits] russell: branch 1.4 r137731 - /branches/1.4/configs/sip.conf.sample
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Thu Aug 14 09:05:24 CDT 2008
Author: russell
Date: Thu Aug 14 09:05:23 2008
New Revision: 137731
URL: http://svn.digium.com/view/asterisk?view=rev&rev=137731
Log:
Comments in this config file were aligned only if your tab size was set to 8.
So, convert tabs to spaces so that things should be aligned regardless of what
tab size you use in your editor.
Modified:
branches/1.4/configs/sip.conf.sample
Modified: branches/1.4/configs/sip.conf.sample
URL: http://svn.digium.com/view/asterisk/branches/1.4/configs/sip.conf.sample?view=diff&rev=137731&r1=137730&r2=137731
==============================================================================
--- branches/1.4/configs/sip.conf.sample (original)
+++ branches/1.4/configs/sip.conf.sample Thu Aug 14 09:05:23 2008
@@ -13,65 +13,65 @@
; where the proxyhostname is defined in a section below
;
; Useful CLI commands to check peers/users:
-; sip show peers Show all SIP peers (including friends)
-; sip show users Show all SIP users (including friends)
-; sip show registry Show status of hosts we register with
-;
-; sip debug Show all SIP messages
-;
-; reload chan_sip.so Reload configuration file
-; Active SIP peers will not be reconfigured
+; sip show peers Show all SIP peers (including friends)
+; sip show users Show all SIP users (including friends)
+; sip show registry Show status of hosts we register with
+;
+; sip debug Show all SIP messages
+;
+; module reload chan_sip.so Reload configuration file
+; Active SIP peers will not be reconfigured
;
[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
@@ -87,50 +87,50 @@
;
;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
@@ -154,23 +154,23 @@
; 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) ----------------------------
@@ -189,24 +189,24 @@
; 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 -----------------------
;
@@ -234,7 +234,7 @@
;
; Examples:
;
-;register => 1234:password at mysipprovider.com
+;register => 1234:password at mysipprovider.com
;
; This will pass incoming calls to the 's' extension
;
@@ -249,34 +249,34 @@
; 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
@@ -285,12 +285,12 @@
; 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
@@ -298,72 +298,72 @@
; 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'
@@ -387,22 +387,22 @@
; 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
@@ -439,8 +439,8 @@
; 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 at digium.com
;
@@ -454,7 +454,7 @@
; User config options: Peer configuration:
; -------------------- -------------------
; context context
-; callingpres callingpres
+; callingpres callingpres
; permit permit
; deny deny
; secret secret
@@ -474,15 +474,15 @@
; 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
@@ -509,25 +509,25 @@
;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
@@ -540,129 +540,129 @@
; 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 at 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
[... 178 lines stripped ...]
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