[DISKUSSION] Asterisk 1.4.0 auf vServer

[SkyTalker]

Hallo nochmal, sorry das so lange nichts gekommen ist...

Hab jetzt Asterisk 1.4.1 und den GUI rev.402 drauf gespielt...
aber aus einem vermutlich verständniss problem will das irgendwie nicht...
hab versucht die 50000 durchzuleiten aber es geht nicht!

Hat irgendjemand mal rein auf der GUI ein und ausgehende rufe zum laufen gebracht ?
Also rein ist kein problem das geht... aber eben raus nicht :-(

was hat es denn mit der extension.ael aufsich ?
ich häng einfach mal meine ganze extension an ps. das ist fast komplet das sample!

;!
;! Automatically generated configuration file
;! Filename: extensions.conf (/etc/asterisk/extensions.conf)
;! Generator: Manager
;! Creation Date: Mon Mar 12 10:56:15 2007
;!
[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.
;
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_1 = SIP/trunk_1
trunk_2 = IAX2/trunk_2
;TRUNK=IAX2/user:pass@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@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,${GLOBAL(CONSOLE)})
exten => 1235,1,Voicemail(1234,u)  ; Right to voicemail
exten => 1236,1,Dial(Console/dsp)  ; Ring forever
exten => 1236,n,Voicemail(1234,b)  ; Unless busy
;
; # for when they're done with the demo
;
exten => #,1,Playback(demo-thanks)  ; "Thanks for trying the demo"
exten => #,n,Hangup  ; Hang them up.
;
; A timeout and "invalid extension rule"
;
exten => t,1,Goto(#,1)  ; If they take too long, give up
exten => i,1,Playback(invalid)  ; "That's not valid, try again"
;
; Create an extension, 500, for dialing the
; Asterisk demo.
;
exten => 500,1,Playback(demo-abouttotry)  ; Let them know what's going on
exten => 500,n,Dial(IAX2/[email protected]/s@default)  ; Call the Asterisk demo
exten => 500,n,Playback(demo-nogo)  ; Couldn't connect to the demo site
exten => 500,n,Goto(s,6)  ; Return to the start over message.
;
; Create an extension, 600, for evaluating echo latency.
;
exten => 600,1,Playback(demo-echotest)  ; Let them know what's going on
exten => 600,n,Echo  ; Do the echo test
exten => 600,n,Playback(demo-echodone)  ; Let them know it's over
exten => 600,n,Goto(s,6)  ; Start over
;
;	You can use the Macro Page to intercom a individual user
exten => 76245,1,Macro(page,SIP/Grandstream1)
; or if your peernames are the same as extensions
exten => _7XXX,1,Macro(page,SIP/${EXTEN})
;
;
; System Wide Page at extension 7999
;
exten => 7999,1,Set(TIMEOUT(absolute)=60)
exten => 7999,2,Page(Local/Grandstream1@page&Local/Xlite1@page&Local/1234@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
exten => 6500,1,VoiceMailMain

[numberplan-custom-1]
plancomment = DialPlan1
include = default
exten = _50000,1,Macro(trunkdial,${trunk_1}/${EXTEN:})
comment = _50000,1,sip mailbox,standard

[asterisk_guitools]
exten = executecommand,1,System(${command})
exten = executecommand,n,Hangup()
exten = record_vmenu,1,Answer
exten = record_vmenu,n,Playback(vm-intro)
exten = record_vmenu,n,Record(${var1})
exten = record_vmenu,n,Playback(vm-saved)
exten = record_vmenu,n,Playback(vm-goodbye)
exten = record_vmenu,n,Hangup
exten = play_file,1,Answer
exten = play_file,n,Playback(${var1})
exten = play_file,n,Hangup
hasbeensetup = N

[DID_trunk_1]
include = default
exten = _X.,1,Goto(default|6000|1)
exten = s,1,Goto(default|6000|1)

[DID_trunk_2]
include = default

Mein Ziel ist es aber das ganze mit dem GUI so hin zu bekommen das es läuft...

 

[Nimrod]

Also ich hab in der 1.4.0 schon ein und ausgehenden SIP Gespräche zum Laufen gebracht, manche Provider erfordern aber ein wenig Eingriff in die Users.conf wo die Providerdaten stehen. Die stehen in der 1.4 übrigens nicht mehr in der sip.conf und register-Befehler werden auch nicht mehr benötigt (ist zumindest meine Erfahrung).

1und1 war einer der wenigen der nachträgliche Bearbeitung brauchte, aber das gabs ja schon immer Schwierigkeiten.

Die 1.4.1 ziert sich sehr bei mir, daher hab ich die nicht am Laufen. Vielleicht bekomme ich das auch noch mal hin. Da ging bei mir nur die Voicemail und eingehende Anrufe, alles andere war tot.

 

[sago]

users.conf

Also dass in * 1.4 nur noch Einträge in users.conf gemacht werden und das
register-directives hinfällig werden, kann ich so nicht sagen.

Ich hab mit

make samples

mir meine anfänglichen config-files erzeugt.

Und da steht folgdender Kommentar in users.conf:

; Using users.conf is not intended to
; provide you with as much flexibility as using the separate configuration
; files (e.g. sip.conf, iax.conf, etc) but is intended to accelerate the
; simple task of adding users. Note that creating individual items (e.g.
; custom SIP peers, IAX friends, etc.) will allow you to override specific
; parameters within this file.

Also:
sip.conf, iax.conf gehen vor und überschreiben Definitionen aus users.conf.

Ich denke users.conf ist ideal, um einfache Tasks über die GUI zentral abzulegen und (noch wichtiger) wieder entfernen zu können. Die volle Flexibilität gibts aber nur mit händischem Eintrag.

Ich verwende users.conf nicht.

Gruß, Stefan

 

[sago]

Wie sichert ihr die GUI ab?

Wie sichert ihr die GUI auf eurem vServer gegen unerwüschte Manipultaion ab?

netfilter?? Passwort??

 

[Nimrod]

Ja das stimmt schon was da über die sip.conf steht. Ich habe die 1.4 mal mit 1.2 context benutzt so wie in betateilchens Anleitung. Das geht ganz gut, man muss nur ein paar Kleinigkeiten ändern weil in 1.4 halt einiges anders ist.

Man kann aber über das GUI die Users und Providers erstellen die dann in die users.conf geschrieben werden. In der extensions landen wie gewohnt die trunks.

Das mit dem netfilter wär ne gute Idee, hab mich auch schon gefragt wie ich sicherstellen kann dass das nur für mich ist. Zertifikat oder ähnliches wär nicht schlecht.

 

[Nimrod]

Installations-Skript hochgeladen

Ich hab gestern meinen Server auf Debian 4.0 umgestellt und musste danach dann mein Asterisk neu installieren. Da ich gerade eh auf der Suche war nach ein paar Anleitungen wie man Gemeinschaft oder FreePBX einfach installieren kann bin ich auf ein Installationsskript gestossen und dachte es wäre eine gute Idee wenn ich das auch mal für meine Version mache.

Man muss einfach nur die Zip-Datei entpacken, drin sind zwei .sh-Dateien die man auf den Server lädt. Die erste installiert alles exakt so wie ich es beschrieben habe, man muss immer nur Enter drück um fortzufahren, man wird also über die einzelnen Schritte informiert. Wen das nicht interessiert der nimmt die Speed-Variante und hat am Ende ein fertig installiertes Asterisk 1.4.19. Man muss die Berechtigungen der Dateien auf 777 setzen, starten geht mit ./asterisk1.4.19.sh bzw ./asterisk1.4.19speed.sh

Vorher muss man ein configure, update und upgrade machen so wie es in den ersten Schritten der Anleitung steht.

Falls es irgendwo zu Fehlern kommen sollte, z.B. bei fehlenden Packages, schreibt das einfach hier rein und ich werde die Skripte so ergänzen dass sie bei mehr Leuten ohne Fehler laufen. Bei Debian 4.0 braucht man so gut wie keine neu installieren.

 

[E.A.V]

sry für faschpost

 

[Guard-X]

Ich hab asterisk 1.4.21.1 auf meinem Suse 10.3 server zuhause laufen.
oder befinde ich mich hier im falschen beitrag?

Wenn ich so die Themenüberschrift lese - ja.

Bitte erstelle doch eine neues Thema...

 

[nudelholz8]

Heyho,

wollte mich auch gerade mal dran machen und auf einem S4U Testserver (für 4 Tage kostenlos) einen Asterisk installieren.

Auf dem vServer ist Debian Linux 4.0 Standard 32-Bit installiert.

Die ersten Schritte der Anleitung haben funktioniert, leider ging es dann aber schon mit einige Fehlern weiter:

cd asterisk-1.4.14
./configure
make
make install
make samples

erste Zeile hat geklappt (ist ja auch nicht so schwer, Ordnerwechsel halt *grins*)

Dann aber ./configure erst ratert er einige Sachen durch, dann kommt aber folgende Meldung:

checking for GNU make... Not Found
configure: error: *** Please install GNU make.  It is required to build Asterisk!

Könnt ihr mir vielleicht sagen, wie man dieses GNU make installiert?
Sorry, aber leider bin ich nicht wirklich ein Linux Profi (sieht man ja *schäm*)...

Wäre super wenn mir jemand dabei helfen könnte.

Vielen Dank schonmal
NH

 

[doxon]

apt-get update
apt-get install make

Ich denke da wird noch viel kommen was du installieren musst

EDIT: Siehe auch diesen Thread: http://www.ip-phone-forum.de/showthread.php?t=141960

 

[sago]

build-essential!

apt-get -y install build-essential

Installiert im wesentlichen das, was du brauchst.

Viel Erfolg, Stefan

 

[nudelholz8]

Vielen Dank an euch beiden,
Installation läuft...

 

[Juscho]

Ich teste das grad mit Asterisk 1.6 und nem vServer bei S4U. Hab Debian Sarge drauf, bei mir kam ein Fehler

configure: error: termcap support not found

den ich mit

apt-get install libncurses-dev

beheben konnte, für alle die es evtl. auch betrifft.

Dann könnte man noch den Pfad zur http.conf angeben, damit man keine falsche löscht.
z.B.

 /etc/asterisk/http.conf

bei mir