[svn-commits] russell: branch 1.6.0 r137783 - in /branches/1.6.0: ./ configs/sip.conf.sample
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Thu Aug 14 10:20:37 CDT 2008
Author: russell
Date: Thu Aug 14 10:20:37 2008
New Revision: 137783
URL: http://svn.digium.com/view/asterisk?view=rev&rev=137783
Log:
Merged revisions 137732 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
................
r137732 | russell | 2008-08-14 09:15:50 -0500 (Thu, 14 Aug 2008) | 12 lines
Merged revisions 137731 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r137731 | russell | 2008-08-14 09:05:23 -0500 (Thu, 14 Aug 2008) | 4 lines
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/1.6.0/ (props changed)
branches/1.6.0/configs/sip.conf.sample
Propchange: branches/1.6.0/
------------------------------------------------------------------------------
Binary property 'trunk-merged' - no diff available.
Modified: branches/1.6.0/configs/sip.conf.sample
URL: http://svn.digium.com/view/asterisk/branches/1.6.0/configs/sip.conf.sample?view=diff&rev=137783&r1=137782&r2=137783
==============================================================================
--- branches/1.6.0/configs/sip.conf.sample (original)
+++ branches/1.6.0/configs/sip.conf.sample Thu Aug 14 10:20:37 2008
@@ -5,51 +5,51 @@
;-----------------------------------------------------------
; In the dialplan (extensions.conf) you can use several
; syntaxes for dialing SIP devices.
-; SIP/devicename
-; SIP/username at domain (SIP uri)
-; SIP/username[:password[:md5secret[:authname[:transport]]]]@host[:port]
-; SIP/devicename/extension
+; SIP/devicename
+; SIP/username at domain (SIP uri)
+; SIP/username[:password[:md5secret[:authname[:transport]]]]@host[:port]
+; SIP/devicename/extension
;
;
; Devicename
-; devicename is defined as a peer in a section below.
+; devicename is defined as a peer in a section below.
;
; username at domain
-; Call any SIP user on the Internet
-; (Don't forget to enable DNS SRV records if you want to use this)
+; Call any SIP user on the Internet
+; (Don't forget to enable DNS SRV records if you want to use this)
;
; devicename/extension
-; If you define a SIP proxy as a peer below, you may call
-; SIP/proxyhostname/user or SIP/user at proxyhostname
-; where the proxyhostname is defined in a section below
-; This syntax also works with ATA's with FXO ports
+; If you define a SIP proxy as a peer below, you may call
+; SIP/proxyhostname/user or SIP/user at proxyhostname
+; where the proxyhostname is defined in a section below
+; This syntax also works with ATA's with FXO ports
;
; SIP/username[:password[:md5secret[:authname]]]@host[:port]
-; This form allows you to specify password or md5secret and authname
-; without altering any authentication data in config.
-; Examples:
-;
-; SIP/*98 at mysipproxy
-; SIP/sales:topsecret::account02 at domain.com:5062
-; SIP/12345678::bc53f0ba8ceb1ded2b70e05c3f91de4f:myname at 192.168.0.1
+; This form allows you to specify password or md5secret and authname
+; without altering any authentication data in config.
+; Examples:
+;
+; SIP/*98 at mysipproxy
+; SIP/sales:topsecret::account02 at domain.com:5062
+; SIP/12345678::bc53f0ba8ceb1ded2b70e05c3f91de4f:myname at 192.168.0.1
;
; All of these dial strings specify the SIP request URI.
; In addition, you can specify a specific To: header by adding an
; exclamation mark after the dial string, like
;
-; SIP/sales at mysipproxy!sales at edvina.net
+; SIP/sales at mysipproxy!sales at edvina.net
;
; CLI Commands
; -------------------------------------------------------------
; 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 set debug Show all SIP messages
-;
-; sip reload 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 set debug Show all SIP messages
+;
+; module reload chan_sip.so Reload configuration file
+; Active SIP peers will not be reconfigured
;
; ** Deprecated configuration options **
@@ -62,24 +62,24 @@
; "setvar" to set variables that can be used in the dialplan for various limits.
[general]
-context=default ; Default context for incoming calls
-;allowguest=no ; Allow or reject guest calls (default is yes)
-;match_auth_username=yes ; if available, match user entry using the
- ; 'username' field from the authentication line
- ; instead of the From: field.
-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 for unencrypted UDP
- ; and TCP sessions is 5060)
- ; bindport is the local UDP port that Asterisk will listen on
-bindaddr=0.0.0.0 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all)
- ; You can specify port here too, like 123.123.123.123:5080
+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.
+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 for unencrypted UDP
+ ; and TCP sessions is 5060)
+ ; bindport is the local UDP port that Asterisk will listen on
+bindaddr=0.0.0.0 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all)
+ ; You can specify port here too, like 123.123.123.123:5080
;
; Note that the TCP and TLS support for chan_sip is currently considered
@@ -88,50 +88,50 @@
; be reflected in this sample configuration file, as well as in the UPGRADE.txt file.
;
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)
+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)
;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.
-
-;tlscertfile=asterisk.pem ; Certificate file (*.pem only) to use for TLS connections
- ; default is to look for "asterisk.pem" in current directory
+ ; 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=asterisk.pem ; Certificate file (*.pem only) to use for TLS connections
+ ; default is to look for "asterisk.pem" in current directory
;tlscafile=</path/to/certificate>
-; If the server your connecting to uses a self signed certificate
-; you should have their certificate installed here so the code can
-; verify the authenticity of their certificate.
+; If the server your connecting to uses a self signed certificate
+; you should have their certificate installed here so the code can
+; verify the authenticity of their certificate.
;tlscadir=</path/to/ca/dir>
-; A directory full of CA certificates. The files must be named with
-; the CA subject name hash value.
-; (see man SSL_CTX_load_verify_locations for more info)
+; A directory full of CA certificates. The files must be named with
+; the CA subject name hash value.
+; (see man SSL_CTX_load_verify_locations for more info)
;tlsdontverifyserver=[yes|no]
-; If set to yes, don't verify the servers certificate when acting as
-; a client. If you don't have the server's CA certificate you can
-; set this and it will connect without requiring tlscafile to be set.
-; Default is no.
+; If set to yes, don't verify the servers certificate when acting as
+; a client. If you don't have the server's CA certificate you can
+; set this and it will connect without requiring tlscafile to be set.
+; Default is no.
;tlscipher=<SSL cipher string>
-; A string specifying which SSL ciphers to use or not use
-; A list of valid SSL cipher strings can be found at:
-; http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS
-
-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")
+; A string specifying which SSL ciphers to use or not use
+; A list of valid SSL cipher strings can be found at:
+; http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS
+
+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 qos.tex or Quality of Service section of asterisk.pdf for a description of these parameters.
;tos_sip=cs3 ; Sets TOS for SIP packets.
@@ -144,24 +144,24 @@
;cos_video=4 ; Sets 802.1p priority for RTP video packets.
;cos_text=3 ; Sets 802.1p priority for RTP text 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
+;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
;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
-;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.
-;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
+;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.
+;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
@@ -178,63 +178,60 @@
;
;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
- ; 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.
-;sdpowner=root ; Allows you to change the username field in the SDP owner string, (o=)
- ; 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.
-;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 (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 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 :-(
-;outboundproxy=proxy.provider.domain ; send outbound signaling to this proxy, not directly to the devices
+;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
+ ; 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.
+;sdpowner=root ; Allows you to change the username field in the SDP owner string, (o=)
+ ; 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.
+;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 (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
+;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 :-(
+;outboundproxy=proxy.provider.domain ; send outbound signaling to this proxy, not directly to the devices
;outboundproxy=proxy.provider.domain:8080 ; send outbound signaling to this proxy, not directly to the devices
;outboundproxy=proxy.provider.domain,force ; Send ALL outbound signalling to proxy, ignoring route: headers
-;outboundproxy=tls://proxy.provider.domain ; same as '=proxy.provider.domain' except we try to connect with tls
+;outboundproxy=tls://proxy.provider.domain ; same as '=proxy.provider.domain' except we try to connect with tls
; ; (could also be tcp,udp) - defining transports on the proxy line only
; ; applies for the global proxy, otherwise use the transport= option
;matchexterniplocally = yes ; Only substitute the externip or externhost setting if it matches
@@ -253,40 +250,40 @@
; separated by '&'. Patterns may be used in regexten.
;
;regcontext=sipregistrations
-;regextenonqualify=yes ; Default "no"
- ; If you have qualify on and the peer becomes unreachable
- ; this setting will enforce inactivation of the regexten
- ; extension for the peer
+;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
;
;--------------------------- SIP timers ----------------------------------------------------
; These timers are used primarily in INVITE transactions.
; The default for Timer T1 is 500 ms or the measured run-trip time between
; Asterisk and the device if you have qualify=yes for the device.
;
-;t1min=100 ; Minimum roundtrip time for messages to monitored hosts
- ; Defaults to 100 ms
-;timert1=500 ; Default T1 timer
- ; 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
+;t1min=100 ; Minimum roundtrip time for messages to monitored hosts
+ ; Defaults to 100 ms
+;timert1=500 ; Default T1 timer
+ ; 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
;--------------------------- RTP timers ----------------------------------------------------
; These timers are currently used for both audio and video streams. The RTP timeouts
; are only applied to the audio channel.
; The settings are settable in the global section as well as per device
;
-;rtptimeout=60 ; Terminate call if 60 seconds of no RTP or RTCP activity
- ; on the audio channel
- ; when we're not on hold. This is to be able to hangup
- ; a call in the case of a phone disappearing from the net,
- ; like a powerloss or grandma tripping over a cable.
-;rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP or RTCP activity
- ; on the audio channel
- ; when we're on hold (must be > rtptimeout)
-;rtpkeepalive=<secs> ; Send keepalives in the RTP stream to keep NAT open
- ; (default is off - zero)
+;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 Session-Timers (RFC 4028)------------------------------------
; SIP Session-Timers provide an end-to-end keep-alive mechanism for active SIP sessions.
@@ -308,28 +305,14 @@
;session-minse=90
;session-refresher=uas
;
-;--------------------------- HASH TABLE SIZES ------------------------------------------------
-; For maximum efficiency, adjust the following
-; values to be slightly larger than the maximum number of in-memory objects (devices).
-; Too large, and space is wasted. Too small, and things will run slower.
-; 563 is probably way too big for small (home) applications, but it
-; should cover most small/medium sites.
-; It is recommended to make the sizes be a prime number!
-; This was internally set to 17 for small-memory applications...
-; All tables default to 563, except when compiled in LOW_MEMORY mode,
-; in which case, they default to 17. You can override this by uncommenting
-; the following, and changing the values.
-;hash_users=563
-;hash_peers=563
-;hash_dialogs=563
;--------------------------- 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) ----------------------------
@@ -350,26 +333,26 @@
; 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.
-;callcounter = yes ; Enable call counters on devices. This can be set per
- ; device too.
-;counteronpeer = yes ; Apply call counting 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 counter instead of applying two call counters,
- ; 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.
+;callcounter = yes ; Enable call counters on devices. This can be set per
+ ; device too.
+;counteronpeer = yes ; Apply call counting 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 counter instead of applying two call counters,
+ ; 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 -----------------------
;
@@ -400,14 +383,14 @@
; A similar effect can be achieved by adding a "callbackextension" option in a peer section.
; this is equivalent to having the following line in the general section:
;
-; register => username:secret at host/callbackextension
+; register => username:secret at host/callbackextension
;
; and more readable because you don't have to write the parameters in two places
; (note that the "port" is ignored - this is a bug that should be fixed).
;
; Examples:
;
-;register => 1234:password at mysipprovider.com
+;register => 1234:password at mysipprovider.com
;
; This will pass incoming calls to the 's' extension
;
@@ -422,11 +405,11 @@
; 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 ------------------------
;
@@ -446,8 +429,8 @@
; Multiple entries are allowed, e.g. a reasonable set is the following:
;
; localnet=192.168.0.0/255.255.0.0 ; RFC 1918 addresses
-; 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 "externally visible" address and port number to be used when talking
@@ -463,9 +446,9 @@
; This approach can be useful if you have a NAT device where you can
; configure the mapping statically. Examples:
;
-; externip = 12.34.56.78 ; use this address.
-; externip = 12.34.56.78:9900 ; use this address and port.
-; externip = mynat.my.org:12600 ; Public address of my nat box.
+; externip = 12.34.56.78 ; use this address.
+; externip = 12.34.56.78:9900 ; use this address and port.
+; externip = mynat.my.org:12600 ; Public address of my nat box.
;
; b. "externhost = hostname[:port]" is similar to "externip" except
; that the hostname is looked up every "externrefresh" seconds
@@ -474,8 +457,8 @@
; Beware, you might suffer from service disruption when the name server
; resolution fails. Examples:
;
-; externhost=foo.dyndns.net ; refreshed periodically
-; externrefresh=180 ; change the refresh interval
+; externhost=foo.dyndns.net ; refreshed periodically
+; externrefresh=180 ; change the refresh interval
;
; c. "stunaddr = stun.server[:port]" queries the STUN server specified
; as an argument to obtain the external address/port.
@@ -483,8 +466,8 @@
; (as a side effect, sending the query also acts as a keepalive for
; the state entry on the nat box):
;
-; stunaddr = foo.stun.com:3478
-; externrefresh = 15
+; stunaddr = foo.stun.com:3478
+; externrefresh = 15
;
; Note that at the moment all these mechanism work only for the SIP socket.
; The IP address discovered with externip/externhost/STUN is reused for
@@ -510,11 +493,11 @@
; However, this is only useful if the external traffic can reach us.
; The following settings are allowed (both globally and in individual sections):
;
-; nat = no ; default. Use NAT mode only according to RFC3581 (;rport)
-; nat = yes ; Always ignore info and assume NAT
-; nat = never ; Never attempt NAT mode or RFC3581 support
-; nat = route ; route = Assume NAT, don't send rport
-; ; (work around more UNIDEN bugs)
+; nat = no ; default. Use NAT mode only according to RFC3581 (;rport)
+; nat = yes ; Always ignore info and assume NAT
+; nat = never ; Never attempt NAT mode or RFC3581 support
+; nat = route ; route = Assume NAT, don't send rport
+; ; (work around more UNIDEN bugs)
;----------------------------------- MEDIA HANDLING --------------------------------
; By default, Asterisk tries to re-invite the audio to an optimal path. If there's
@@ -522,72 +505,72 @@
; 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.
-
- ; 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.
+
+ ; 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 ------------------------
[... 493 lines stripped ...]
More information about the svn-commits
mailing list