[Asterisk-video] Confiance video mixer
Takashi Ohashi
ohashi at fko.it-tokyo.co.jp
Thu Sep 13 04:31:24 CDT 2007
Hi Lorenzo
I appreciate your response soon.
It as soon as I'll step in there
> 1) check that the video device in the Minisip preferences is correctly
setup
Now I think that the video device in the Minisip preferences is correctly
setup.
> 2) check that the videomixer settings are correctly set in the xcon.conf
file
I attach xcon.conf and extensions.conf below.
> 3) if so, do you at least see the Confiance logo when joining a
conference?
When joining a conference, I can not see the Confiance logo.
---------------xcon.conf------------------------------
[general]
prefix => 867 ; All conference numbers will start with
this number
maxconf => 10 ; We allow at max maxconf conferences at a
time
bfcp_port => 2345 ; BFCP Server (or Gateway, if enabled) will
listen on this port
bfcp_transport => TCP; ; Both TCP/BFCP (TCP) and TCP/TLS/BFCP (TLS)
are supported
scheduler_port => 2346 ; Scheduling Server will listen on this port
debug => 3 ; Debug level for console verbose output:
; 0 - no debug (only joins/leaves)
; 1 - only DCON/XCON messages
; 2 - BFCP messages too
; 3 - full debug
; 4 - VERY full debug (also useless
MSRP log)
;[tls] ; Certificate and PrivateKey files needed by
TLS, if enabled
;certificate => /path/to/server.pem
;privatekey => /path/to/server.key
;[msrp] ; Settings for MSRP, if used
;msrp => yes ; Whether MSRP support must be enabled
;msrp_ip => 127.0.0.1 ; The address to report in MSRP negotiations
[videomixer] ; Settings for the external VideoMixer, if
available
mcu => cvm ; The type of available VideoMixer:
; no = no Videomixer (just basic
videoswitching)
; cvm = Confiance VideoMixer
; murillo = Sergio Garcia Murillo's
VideoMixer
mcu_ip => 127.0.0.1 ; The IP the VideoMixer is on...
mcu_port => 7000 ; ...and the port it is bound to,
defaults are:
; 7000 for cvm
; 8080 for murillo
;mcu_ownip => 127.0.0.1; ; The IP Asterisk is on, needed by
S.G.Murillo's VideoMixer
; to start the transmission of video frames
[manager]
admin => manager ; The admin's username and password will
have to be the same as the Asterisk
secret => mypassword ; Manager Interface account used to get the
XCON events (needed for the Update)
[rooms]
conf => 8671000 ; Create a new room, 8671000 (no phone PIN,
public conference/free access)
[bfcp]
; Conference 8671000 has 'My default XCON
conference' as topic
subject => 8671000,My default XCON conference
password => 8671000,1234 ; Conference 8671000 has 1234 as XconMe
Administrator numeric password
policy => 8671000,2,1,accept ; Conference 8671000 policies are:
; - support 2 floors at max
; - accept 1 floor request per floor
at a time by users
; - auto-accept requests for floors
with no chair
; Conference 8671000 will have these floors:
floor => 8671000,audio,11,0 ; - Floor 11 is Audio, and can be
granted to an unlimited number of users at a time
floor => 8671000,video,22,4 ; - Floor 22 is Video, and can only be
granted to 4 users at a time
(change video users from 4
to 1 if there's no VideoMixer)
-------------------------------------------------
-----------------------extensions.conf-----------
[general]
prefix => 867 ; All conference numbers will start with
this number
maxconf => 10 ; We allow at max maxconf conferences at a
time
bfcp_port => 2345 ; BFCP Server (or Gateway, if enabled) will
listen on this port
bfcp_transport => TCP; ; Both TCP/BFCP (TCP) and TCP/TLS/BFCP (TLS)
are supported
scheduler_port => 2346 ; Scheduling Server will listen on this port
debug => 3 ; Debug level for console verbose output:
; 0 - no debug (only joins/leaves)
; 1 - only DCON/XCON messages
; 2 - BFCP messages too
; 3 - full debug
; 4 - VERY full debug (also useless
MSRP log)
;[tls] ; Certificate and PrivateKey files needed by
TLS, if enabled
;certificate => /path/to/server.pem
;privatekey => /path/to/server.key
;[msrp] ; Settings for MSRP, if used
;msrp => yes ; Whether MSRP support must be enabled
;msrp_ip => 127.0.0.1 ; The address to report in MSRP negotiations
[videomixer] ; Settings for the external VideoMixer, if
available
mcu => cvm ; The type of available VideoMixer:
; no = no Videomixer (just basic
videoswitching)
; cvm = Confiance VideoMixer
; murillo = Sergio Garcia Murillo's
VideoMixer
mcu_ip => 127.0.0.1 ; The IP the VideoMixer is on...
mcu_port => 7000 ; ...and the port it is bound to,
defaults are:
; 7000 for cvm
; 8080 for murillo
;mcu_ownip => 127.0.0.1; ; The IP Asterisk is on, needed by
S.G.Murillo's VideoMixer
; to start the transmission of video frames
[manager]
admin => manager ; The admin's username and password will
have to be the same as the Asterisk
secret => mypassword ; Manager Interface account used to get the
XCON events (needed for the Update)
[rooms]
conf => 8671000 ; Create a new room, 8671000 (no phone PIN,
public conference/free access)
[bfcp]
; Conference 8671000 has 'My default XCON
conference' as topic
subject => 8671000,My default XCON conference
password => 8671000,1234 ; Conference 8671000 has 1234 as XconMe
Administrator numeric password
policy => 8671000,2,1,accept ; Conference 8671000 policies are:
; - support 2 floors at max
; - accept 1 floor request per floor
at a time by users
; - auto-accept requests for floors
with no chair
; Conference 8671000 will have these floors:
floor => 8671000,audio,11,0 ; - Floor 11 is Audio, and can be
granted to an unlimited number of users at a time
floor => 8671000,video,22,4 ; - Floor 22 is Video, and can only be
granted to 4 users at a time
(change video users from 4
to 1 if there's no VideoMixer)
[root at smigmx6 ~]# cat /etc/asterisk/extensions.conf
; extensions.conf - the Asterisk dial plan
;
; Static extension configuration file, used by
; the pbx_config module. This is where you configure all your
; inbound and outbound calls in Asterisk.
;
; This configuration file is reloaded
; - With the "extensions reload" command in the CLI
; - With the "reload" command (that reloads everything) in the CLI
;
; The "General" category is for certain variables.
;
[general]
;
; If static is set to no, or omitted, then the pbx_config will rewrite
; this file when extensions are modified. Remember that all comments
; made in the file will be lost when that happens.
;
; XXX Not yet implemented XXX
;
static=yes
;
; if static=yes and writeprotect=no, you can save dialplan by
; CLI command 'save dialplan' too
;
writeprotect=no
;
; If autofallthrough is set, then if an extension runs out of
; things to do, it will terminate the call with BUSY, CONGESTION
; or HANGUP depending on Asterisk's best guess. This is the default.
;
; If autofallthrough is not set, then if an extension runs out of
; things to do, Asterisk will wait for a new extension to be dialed
; (this is the original behavior of Asterisk 1.0 and earlier).
;
;autofallthrough=no
;
; If clearglobalvars is set, global variables will be cleared
; and reparsed on an extensions reload, or Asterisk reload.
;
; If clearglobalvars is not set, then global variables will persist
; through reloads, and even if deleted from the extensions.conf or
; one of its included files, will remain set to the previous value.
;
; NOTE: A complication sets in, if you put your global variables into
; the AEL file, instead of the extensions.conf file. With clearglobalvars
; set, a 'reload' will often leave the globals vars cleared, because it
; is not unusual to have extensions.conf (which will have no globals)
; load after the extensions.ael file (where the global vars are stored).
; So, with 'reload' in this particular situation, first the AEL file will
; clear and then set all the global vars, then, later, when the
extensions.conf
; file is loaded, the global vars are all cleared, and then not set, because
; they are not stored in the extensions.conf file.
;
clearglobalvars=no
;
; If priorityjumping is set to 'yes', then applications that support
; 'jumping' to a different priority based on the result of their operations
; will do so (this is backwards compatible behavior with pre-1.2 releases
; of Asterisk). Individual applications can also be requested to do this
; by passing a 'j' option in their arguments.
;
;priorityjumping=yes
;
; User context is where entries from users.conf are registered. The
; default value is 'default'
;
;userscontext=default
;
; You can include other config files, use the #include command
; (without the ';'). Note that this is different from the "include" command
; that includes contexts within other contexts. The #include command works
; in all asterisk configuration files.
;#include "filename.conf"
; The "Globals" category contains global variables that can be referenced
; in the dialplan with the GLOBAL dialplan function:
; ${GLOBAL(VARIABLE)}
; ${${GLOBAL(VARIABLE)}} or ${text${GLOBAL(VARIABLE)}} or any hybrid
; Unix/Linux environmental variables can be reached with the ENV dialplan
; function: ${ENV(VARIABLE)}
;
[globals]
CONSOLE=Console/dsp ; Console interface for demo
;CONSOLE=Zap/1
;CONSOLE=Phone/phone0
IAXINFO=guest ; IAXtel username/password
;IAXINFO=myuser:mypass
TRUNK=Zap/g2 ; Trunk interface
;
; Note the 'g2' in the TRUNK variable above. It specifies which group
(defined
; in zapata.conf) to dial, i.e. group 2, and how to choose a channel to use
in
; the specified group. The four possible options are:
;
; g: select the lowest-numbered non-busy Zap channel
; (aka. ascending sequential hunt group).
; G: select the highest-numbered non-busy Zap channel
; (aka. descending sequential hunt group).
; r: use a round-robin search, starting at the next highest channel than
last
; time (aka. ascending rotary hunt group).
; R: use a round-robin search, starting at the next lowest channel than last
; time (aka. descending rotary hunt group).
;
TRUNKMSD=1 ; MSD digits to strip
(usually 1 or 0)
;TRUNK=IAX2/user:pass at provider
;
; Any category other than "General" and "Globals" represent
; extension contexts, which are collections of extensions.
;
; Extension names may be numbers, letters, or combinations
; thereof. If an extension name is prefixed by a '_'
; character, it is interpreted as a pattern rather than a
; literal. In patterns, some characters have special meanings:
;
; X - any digit from 0-9
; Z - any digit from 1-9
; N - any digit from 2-9
; [1235-9] - any digit in the brackets (in this example, 1,2,3,5,6,7,8,9)
; . - wildcard, matches anything remaining (e.g. _9011. matches
; anything starting with 9011 excluding 9011 itself)
; ! - wildcard, causes the matching process to complete as soon as
; it can unambiguously determine that no other matches are possible
;
; For example the extension _NXXXXXX would match normal 7 digit dialings,
; while _1NXXNXXXXXX would represent an area code plus phone number
; preceded by a one.
;
; Each step of an extension is ordered by priority, which must
; always start with 1 to be considered a valid extension. The priority
; "next" or "n" means the previous priority plus one, regardless of whether
; the previous priority was associated with the current extension or not.
; The priority "same" or "s" means the same as the previously specified
; priority, again regardless of whether the previous entry was for the
; same extension. Priorities may be immediately followed by a plus sign
; and another integer to add that amount (most useful with 's' or 'n').
; Priorities may then also have an alias, or label, in
; parenthesis after their name which can be used in goto situations
;
; Contexts contain several lines, one for each step of each
; extension, which can take one of two forms as listed below,
; with the first form being preferred. One may include another
; context in the current one as well, optionally with a
; date and time. Included contexts are included in the order
; they are listed.
;
;[context]
;exten =>
someexten,{priority|label{+|-}offset}[(alias)],application(arg1,arg2,...)
;exten =>
someexten,{priority|label{+|-}offset}[(alias)],application,arg1|arg2...
;
; Timing list for includes is
;
; <time range>|<days of week>|<days of month>|<months>
;
; Note that ranges may be specified to wrap around the ends. Also, minutes
are
; fine-grained only down to the closest even minute.
;
;include => daytime|9:00-17:00|mon-fri|*|*
;include => weekend|*|sat-sun|*|*
;include => weeknights|17:02-8:58|mon-fri|*|*
;
; ignorepat can be used to instruct drivers to not cancel dialtone upon
; receipt of a particular pattern. The most commonly used example is
; of course '9' like this:
;
;ignorepat => 9
;
; so that dialtone remains even after dialing a 9.
;
;
; Sample entries for extensions.conf
;
;
[dundi-e164-canonical]
;
; List canonical entries here
;
;exten => 12564286000,1,Macro(stdexten,6000,IAX2/foo)
;exten => _125642860XX,1,Dial(IAX2/otherbox/${EXTEN:7})
[dundi-e164-customers]
;
; If you are an ITSP or Reseller, list your customers here.
;
;exten => _12564286000,1,Dial(SIP/customer1)
;exten => _12564286001,1,Dial(IAX2/customer2)
[dundi-e164-via-pstn]
;
; If you are freely delivering calls to the PSTN, list them here
;
;exten => _1256428XXXX,1,Dial(Zap/g2/${EXTEN:7}) ; Expose all of 256-428
;exten => _1256325XXXX,1,Dial(Zap/g2/${EXTEN:7}) ; Ditto for 256-325
[dundi-e164-local]
;
; Context to put your dundi IAX2 or SIP user in for
; full access
;
include => dundi-e164-canonical
include => dundi-e164-customers
include => dundi-e164-via-pstn
[dundi-e164-switch]
;
; Just a wrapper for the switch
;
switch => DUNDi/e164
[dundi-e164-lookup]
;
; Locally to lookup, try looking for a local E.164 solution
; then try DUNDi if we don't have one.
;
include => dundi-e164-local
include => dundi-e164-switch
;
; DUNDi can also be implemented as a Macro instead of using
; the Local channel driver.
;
[macro-dundi-e164]
;
; ARG1 is the extension to Dial
;
; Extension "s" is not a wildcard extension that matches "anything".
; In macros, it is the start extension. In most other cases,
; you have to goto "s" to execute that extension.
;
; For wildcard matches, see above - all pattern matches start with
; an underscore.
exten => s,1,Goto(${ARG1},1)
include => dundi-e164-lookup
;
; Here are the entries you need to participate in the IAXTEL
; call routing system. Most IAXTEL numbers begin with 1-700, but
; there are exceptions. For more information, and to sign
; up, please go to www.gnophone.com or www.iaxtel.com
;
[iaxtel700]
exten =>
_91700XXXXXXX,1,Dial(IAX2/${GLOBAL(IAXINFO)}@iaxtel.com/${EXTEN:1}@iaxtel)
;
; The SWITCH statement permits a server to share the dialplan with
; another server. Use with care: Reciprocal switch statements are not
; allowed (e.g. both A -> B and B -> A), and the switched server needs
; to be on-line or else dialing can be severly delayed.
;
[iaxprovider]
;switch => IAX2/user:[key]@myserver/mycontext
[trunkint]
;
; International long distance through trunk
;
exten => _9011.,1,Macro(dundi-e164,${EXTEN:4})
exten => _9011.,n,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
[trunkld]
;
; Long distance context accessed through trunk
;
exten => _91NXXNXXXXXX,1,Macro(dundi-e164,${EXTEN:1})
exten => _91NXXNXXXXXX,n,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
[trunklocal]
;
; Local seven-digit dialing accessed through trunk interface
;
exten => _9NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
[trunktollfree]
;
; Long distance context accessed through trunk interface
;
exten => _91800NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
exten => _91888NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
exten => _91877NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
exten => _91866NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
[international]
;
; Master context for international long distance
;
ignorepat => 9
include => longdistance
include => trunkint
[longdistance]
;
; Master context for long distance
;
ignorepat => 9
include => local
include => trunkld
[local]
;
; Master context for local, toll-free, and iaxtel calls only
;
ignorepat => 9
include => default
include => trunklocal
include => iaxtel700
include => trunktollfree
include => iaxprovider
;Include parkedcalls (or the context you define in features conf)
;to enable call parking.
include => parkedcalls
;
; You can use an alternative switch type as well, to resolve
; extensions that are not known here, for example with remote
; IAX switching you transparently get access to the remote
; Asterisk PBX
;
; switch => IAX2/user:password at bigserver/local
;
; An "lswitch" is like a switch but is literal, in that
; variable substitution is not performed at load time
; but is passed to the switch directly (presumably to
; be substituted in the switch routine itself)
;
; lswitch => Loopback/12${EXTEN}@othercontext
;
; An "eswitch" is like a switch but the evaluation of
; variable substitution is performed at runtime before
; being passed to the switch routine.
;
; eswitch => IAX2/context@${CURSERVER}
[macro-trunkdial]
;
; Standard trunk dial macro (hangs up on a dialstatus that should
; terminate call)
; ${ARG1} - What to dial
;
exten => s,1,Dial(${ARG1})
exten => s,n,Goto(s-${DIALSTATUS},1)
exten => s-NOANSWER,1,Hangup
exten => s-BUSY,1,Hangup
exten => _s-.,1,NoOp
[macro-stdexten];
;
; Standard extension macro:
; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as well
; ${ARG2} - Device(s) to ring
;
exten => s,1,Dial(${ARG2},20) ; Ring the interface, 20
seconds maximum
exten => s,2,Goto(s-${DIALSTATUS},1) ; Jump based on status
(NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten => s-NOANSWER,1,Voicemail(${ARG1},u) ; If unavailable, send to
voicemail w/ unavail announce
exten => s-NOANSWER,2,Goto(default,s,1) ; If they press #, return to
start
exten => s-BUSY,1,Voicemail(${ARG1},b) ; If busy, send to voicemail
w/ busy announce
exten => s-BUSY,2,Goto(default,s,1) ; If they press #, return to
start
exten => _s-.,1,Goto(s-NOANSWER,1) ; Treat anything else as no
answer
exten => a,1,VoicemailMain(${ARG1}) ; If they press *, send the
user into VoicemailMain
[macro-stdPrivacyexten];
;
; Standard extension macro:
; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as well
; ${ARG2} - Device(s) to ring
; ${ARG3} - Optional DONTCALL context name to jump to (assumes the s,1
extension-priority)
; ${ARG4} - Optional TORTURE context name to jump to (assumes the s,1
extension-priority)`
;
exten => s,1,Dial(${ARG2},20|p) ; Ring the interface, 20
seconds maximum, call screening
; option (or use P for
databased call screening)
exten => s,2,Goto(s-${DIALSTATUS},1) ; Jump based on status
(NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten => s-NOANSWER,1,Voicemail(${ARG1},u) ; If unavailable, send to
voicemail w/ unavail announce
exten => s-NOANSWER,2,Goto(default,s,1) ; If they press #, return to
start
exten => s-BUSY,1,Voicemail(${ARG1},b) ; If busy, send to voicemail
w/ busy announce
exten => s-BUSY,2,Goto(default,s,1) ; If they press #, return to
start
exten => s-DONTCALL,1,Goto(${ARG3},s,1) ; Callee chose to send this
call to a polite "Don't call again" script.
exten => s-TORTURE,1,Goto(${ARG4},s,1) ; Callee chose to send this
call to a telemarketer torture script.
exten => _s-.,1,Goto(s-NOANSWER,1) ; Treat anything else as no
answer
exten => a,1,VoicemailMain(${ARG1}) ; If they press *, send the
user into VoicemailMain
[macro-page];
;
; Paging macro:
;
; Check to see if SIP device is in use and DO NOT PAGE if they are
;
; ${ARG1} - Device to page
exten => s,1,ChanIsAvail(${ARG1}|js) ; j is for Jump and
s is for ANY call
exten => s,n,GoToIf([${AVAILSTATUS} = "1"]?autoanswer:fail)
exten => s,n(autoanswer),Set(_ALERT_INFO="RA") ; This is
for the PolyComs
exten => s,n,SIPAddHeader(Call-Info: Answer-After=0) ; This is for the
Grandstream, Snoms, and Others
exten => s,n,NoOp() ; Add others here
and Post on the Wiki!!!!
exten => s,n,Dial(${ARG1}||)
exten => s,n(fail),Hangup
[demo]
;
; We start with what to do when a call first comes in.
;
exten => s,1,Wait(1) ; Wait a second, just for fun
exten => s,n,Answer ; Answer the line
exten => s,n,Set(TIMEOUT(digit)=5) ; Set Digit Timeout to 5 seconds
exten => s,n,Set(TIMEOUT(response)=10) ; Set Response Timeout to 10 seconds
exten => s,n(restart),BackGround(demo-congrats) ; Play a congratulatory
message
exten => s,n(instruct),BackGround(demo-instruct) ; Play some
instructions
exten => s,n,WaitExten ; Wait for an extension to be
dialed.
exten => 2,1,BackGround(demo-moreinfo) ; Give some more information.
exten => 2,n,Goto(s,instruct)
exten => 3,1,Set(LANGUAGE()=fr) ; Set language to french
exten => 3,n,Goto(s,restart) ; Start with the congratulations
exten => 1000,1,Goto(default,s,1)
;
; We also create an example user, 1234, who is on the console and has
; voicemail, etc.
;
exten => 1234,1,Playback(transfer,skip) ; "Please hold while..."
; (but skip if channel is not up)
exten => 1234,n,Macro(stdexten,1234,${GLOBAL(CONSOLE)})
exten => 1235,1,Voicemail(1234,u) ; Right to voicemail
exten => 1236,1,Dial(Console/dsp) ; Ring forever
exten => 1236,n,Voicemail(1234,b) ; Unless busy
;
; # for when they're done with the demo
;
exten => #,1,Playback(demo-thanks) ; "Thanks for trying the demo"
exten => #,n,Hangup ; Hang them up.
;
; A timeout and "invalid extension rule"
;
exten => t,1,Goto(#,1) ; If they take too long, give up
exten => i,1,Playback(invalid) ; "That's not valid, try again"
;
; Create an extension, 500, for dialing the
; Asterisk demo.
;
exten => 500,1,Playback(demo-abouttotry); Let them know what's going on
exten => 500,n,Dial(IAX2/guest at misery.digium.com/s at default) ; Call the
Asterisk demo
exten => 500,n,Playback(demo-nogo) ; Couldn't connect to the demo site
exten => 500,n,Goto(s,6) ; Return to the start over message.
;
; Create an extension, 600, for evaluating echo latency.
;
exten => 600,1,Playback(demo-echotest) ; Let them know what's going on
exten => 600,n,Echo ; Do the echo test
exten => 600,n,Playback(demo-echodone) ; Let them know it's over
exten => 600,n,Goto(s,6) ; Start over
;
; You can use the Macro Page to intercom a individual user
exten => 76245,1,Macro(page,SIP/Grandstream1)
; or if your peernames are the same as extensions
exten => _7XXX,1,Macro(page,SIP/${EXTEN})
;
;
; System Wide Page at extension 7999
;
exten => 7999,1,Set(TIMEOUT(absolute)=60)
exten =>
7999,2,Page(Local/Grandstream1 at page&Local/Xlite1 at page&Local/1234 at page/n|d)
; Give voicemail at extension 8500
;
exten => 8500,1,VoicemailMain
exten => 8500,n,Goto(s,6)
;
; Here's what a phone entry would look like (IXJ for example)
;
;exten => 1265,1,Dial(Phone/phone0,15)
;exten => 1265,n,Goto(s,5)
;
; The page context calls up the page macro that sets variables needed
for auto-answer
; It is in is own context to make calling it from the Page()
application as simple as
; Local/{peername}@page
;
[page]
exten => _X.,1,Macro(page,SIP/${EXTEN})
;[mainmenu]
;
; Example "main menu" context with submenu
;
;exten => s,1,Answer
;exten => s,n,Background(thanks) ; "Thanks for calling press
1 for sales, 2 for support, ..."
;exten => s,n,WaitExten
;exten => 1,1,Goto(submenu,s,1)
;exten => 2,1,Hangup
;include => default
;
;[submenu]
;exten => s,1,Ringing ; Make them
comfortable with 2 seconds of ringback
;exten => s,n,Wait,2
;exten => s,n,Background(submenuopts) ; "Thanks for calling the sales
department. Press 1 for steve, 2 for..."
;exten => s,n,WaitExten
;exten => 1,1,Goto(default,steve,1)
;exten => 2,1,Goto(default,mark,2)
[default]
;
; By default we include the demo. In a production system, you
; probably don't want to have the demo there.
;
include => demo
;
; An extension like the one below can be used for FWD, Nikotel, sipgate etc.
; Note that you must have a [sipprovider] section in sip.conf
;
;exten => _41X.,1,Dial(SIP/${EXTEN:2}@sipprovider,,r)
; Real extensions would go here. Generally you want real extensions to be
; 4 or 5 digits long (although there is no such requirement) and start with
a
; single digit that is fairly large (like 6 or 7) so that you have plenty of
; room to overlap extensions and menu options without conflict. You can
alias
; them with names, too, and use global variables
;exten => 6245,hint,SIP/Grandstream1&SIP/Xlite1,Joe Schmoe ; Channel hints
for presence
;exten => 6245,1,Dial(SIP/Grandstream1,20,rt) ; permit transfer
;exten => 6245,n(dial),Dial(${HINT},20,rtT) ; Use hint as listed
;exten => 6245,n,Voicemail(6245,u) ; Voicemail (unavailable)
;exten => 6245,s+1,Hangup ; s+1, same as n
;exten => 6245,dial+101,Voicemail(6245,b) ; Voicemail (busy)
;exten => 6361,1,Dial(IAX2/JaneDoe,,rm) ; ring without time limit
;exten => 6389,1,Dial(MGCP/aaln/1 at 192.168.0.14)
;exten => 6390,1,Dial(JINGLE/caller/callee) ; Dial via jingle using labels
;exten => 6391,1,Dial(JINGLE/asterisk at digium.com/mogorman at astjab.org) ;Dial
via jingle using asterisk as the transport and calling mogorman.
;exten => 6394,1,Dial(Local/6275/n) ; this will dial ${MARK}
;exten => 6275,1,Macro(stdexten,6275,${MARK}) ; assuming ${MARK} is
something like Zap/2
;exten => mark,1,Goto(6275|1) ; alias mark to 6275
;exten => 6536,1,Macro(stdexten,6236,${WIL}) ; Ditto for wil
;exten => wil,1,Goto(6236|1)
;If you want to subscribe to the status of a parking space, this is
;how you do it. Subscribe to extension 6600 in sip, and you will see
;the status of the first parking lot with this extensions' help
;exten => 6600,hint,park:701 at parkedcalls
;exten => 6600,1,noop
;
; Some other handy things are an extension for checking voicemail via
; voicemailmain
;
;exten => 8500,1,VoicemailMain
;exten => 8500,n,Hangup
;
; Or a conference room (you'll need to edit meetme.conf to enable this room)
;
;exten => 8600,1,Meetme(1234)
;
; Or playing an announcement to the called party, as soon it answers
;
exten = 8700,1,Dial(${MARK},30,A(/path/to/my/announcemsg))
;
;
; XCON through MeetMe: example of wildcards to add flexibility
; - First 7 numbers = conference
; - Next (1-4) numbers = PIN (Phone PIN, not Admin's password)
;
; the 'B' flag tells MeetMe this is an XCON conference (B => BFCP)
;
exten => _867.,1,Meetme(${EXTEN:0:7}|B|${EXTEN:7})
exten => _867.,2,Hangup
;
; XconScheduler
;
exten => 868,1,Answer
exten => 868,2,XconScheduler()
exten => 868,3,Hangup
;
;
; For more information on applications, just type "show applications" at
your
; friendly Asterisk CLI prompt.
;
; 'show application <command>' will show details of how you
; use that particular application in this file, the dial plan.
; 'show functions" will list all dialplan functions
; 'show function <COMMAND>' will show you more information about
; one function. Remember that function names are UPPER CASE.
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