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[gelöst] Fehlender Ton

Dieses Thema im Forum "Asterisk ISDN mit CAPI (chan_capi, chan_capi_cm)" wurde erstellt von topher, 23 Aug. 2005.

  1. topher

    topher Neuer User

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    Hi zusammen,

    nachdem ich nun das Problem mit dem Installieren des asterisk und des chan_capi gelöst habe. Schaffe ich es leider nicht die einzelnen Dateien richtig einzurichten.

    [Edit]
    Also.. das Ganze klappt jetz und wir können auch raustelefonieren. Das einzige Problem ist noch.. dass der vom Computer auf Festnetz anruft gehört wird, aber der, der vom Telefon spricht nicht ankommt. Woran kann das liegen, dass die Tonübermittelung fehlt?

    modules.conf
    Code:
    ;
    ; Asterisk configuration file
    ;
    ; Module Loader configuration file
    ;
    
    [modules]
    autoload=yes
    ;
    ; If you want, load the GTK console right away.  
    ; Don't load the KDE console since
    ; it's not as sophisticated right now.
    ;
    noload => pbx_gtkconsole.so
    ;load => pbx_gtkconsole.so
    noload => pbx_kdeconsole.so
    ;
    ; Intercom application is obsoleted by
    ; chan_oss.  Don't load it.
    ;
    noload => app_intercom.so
    ;
    ; Explicitly load the chan_modem.so early on to be sure
    ; it loads before any of the chan_modem_* 's afte rit
    ;
    noload => chan_modem.so
    noload => chan_modem_aopen.so
    noload => chan_modem_i4l.so
    noload => chan_modem_bestdata.so
    load => chan_capi.so
    load => res_features.so
    load => res_musiconhold.so
    ;
    ; Load either OSS or ALSA, not both
    ; By default, load OSS only (automatically) and do not load ALSA
    ;
    noload => chan_alsa.so
    ;noload => chan_oss.so
    ;
    ; Module names listed in "global" section will have symbols globally
    ; exported to modules loaded after them.
    ;
    [global]
    chan_capi.so=yes
    chan_modem.so=no
    iax.conf
    Code:
    
    ; Inter-Asterisk eXchange driver definition
    ;
    ;
    ; General settings, like port number to bind to, and
    ; an option address (the default is to bind to all
    ; local addresses).
    ;
    [general]
    ;bindport=4569
    ;bindaddr=192.168.0.1
    ;
    ; Set iaxcompat to yes if you plan to use layered 
    ; switches.  It incurs a small performance hit to enable it
    ;
    ;iaxcompat=yes
    ;
    ; For increased security against brute force password attacks
    ; enable "delayreject" which will delay the sending of authentication
    ; reject for REGREQ or AUTHREP if there is a password.  
    ;
    ;delayreject=yes
    ;
    ; You may specify a global default AMA flag for iaxtel calls.  It must be
    ; one of 'default', 'omit', 'billing', or 'documentation'.  These flags
    ; are used in the generation of call detail records.
    ;
    ;amaflags=default
    ;
    ; You may specify a default account for Call Detail Records in addition
    ; to specifying on a per-user basis
    ;
    ;accountcode=lss0101
    ;
    ; You may specify a global default language for users. 
    ; Can be specified also on a per-user basis
    ; If omitted, will fallback to english
    ;
    ;language=en
    ;
    ; Specify bandwidth of low, medium, or high to control which codecs are used
    ; in general.
    ;
    bandwidth=low
    ;
    ; You can also fine tune codecs here using "allow" and "disallow" clauses
    ; with specific codecs.  Use "all" to represent all formats.
    ;
    ;allow=all			; same as bandwidth=high
    ;disallow=g723.1		; Hm...  Proprietary, don't use it...
    disallow=lpc10			; Icky sound quality...  Mr. Roboto.
    ;allow=gsm			; Always allow GSM, it's cool :)
    ;
    
    ; You can adjust several parameters relating to the jitter buffer.
    ; The jitter buffer's function is to compensate for varying
    ; network delay.
    ;
    ; All the jitter buffer settings except dropcount are in milliseconds.
    ; The jitter buffer works for INCOMING audio - the outbound audio
    ; will be dejittered by the jitter buffer at the other end.
    ;
    ; jitterbuffer=yes|no: global default as to whether you want
    ; the jitter buffer at all.
    ;
    ; dropcount: the jitter buffer is sized such that no more than "dropcount"
    ; frames would have been "too late" over the last 2 seconds.
    ; Set to a small number.  "3" represents 1.5% of frames dropped
    ;
    ; maxjitterbuffer: a maximum size for the jitter buffer.
    ; Setting a reasonable maximum here will prevent the call delay
    ; from rising to silly values in extreme situations; you'll hear
    ; SOMETHING, even though it will be jittery.
    ;
    ; maxexcessbuffer: If conditions improve after a period of high jitter,
    ; the jitter buffer can end up bigger than necessary.  If it ends up
    ; more than "maxexcessbuffer" bigger than needed, Asterisk will start
    ; gradually decreasing the amount of jitter buffering.
    ;
    ; minexcessbuffer: Sets a desired mimimum amount of headroom in 
    ; the jitter buffer.  If Asterisk has less headroom than this, then
    ; it will start gradually increasing the amount of jitter buffering.
    ;
    ; jittershrinkrate: when the jitter buffer is being gradually shrunk 
    ; (or enlarged), how many millisecs shall we take off per 20ms frame
    ; received?  Use a small number, or you will be able to hear it
    ; changing.  An example: if you set this to 2, then the jitter buffer
    ; size will change by 100 millisecs per second.
    
    jitterbuffer=no
    ;dropcount=2
    ;maxjitterbuffer=500
    ;maxexcessbuffer=80
    ;minexcessbuffer=10
    ;jittershrinkrate=1
    
    ;trunkfreq=20			; How frequently to send trunk msgs (in ms)
    ;
    ;
    ; We can register with another IAX server to let him know where we are
    ; in case we have a dynamic IP address for example
    ;
    ; Register with tormenta using username marko and password secretpass
    ;
    ;register => marko:secretpass@tormenta.linux-support.net
    ;
    ; Register joe at remote host with no password
    ;
    ;register => joe@remotehost:5656
    ;
    ; Register marko at tormenta.linux-support.net using RSA key "torkey"
    ;
    ;register => marko:[torkey]@tormenta.linux-support.net
    ;
    ; Sample Registration for iaxtel
    ;
    ; Visit http://www.iaxtel.com to register with iaxtel.  Replace "user"
    ; and "pass" with your username and password for iaxtel.  Incoming 
    ; calls arrive at the "s" extension of "default" context.
    ;
    ;register => user:pass@iaxtel.com
    ;
    ; Sample Registration for IAX + FWD
    ;
    ; To register using IAX with FWD, it must be enabled by visiting the URL
    ; http://www.fwdnet.net/index.php?section_id=112
    ;
    ; Note that you need an extension in you default context which matches
    ; your free world dialup number.  Please replace "FWDNumber" with your
    ; FWD number and "passwd" with your password.
    ;
    ;register => FWDNumber:passwd@iax.fwdnet.net
    ;
    ;
    ; You can disable authentication debugging to reduce the amount of 
    ; debugging traffic.
    ;
    ;authdebug=no
    ;
    ; Finally, you can set values for your TOS bits to help improve 
    ; performance.  Valid values are:
    ;   lowdelay		-- Minimize delay
    ;   throughput		-- Maximize throughput
    ;   reliability		-- Maximize reliability
    ;   mincost		-- Minimize cost
    ;   none		-- No flags
    ;
    tos=lowdelay
    ;
    ; If mailboxdetail is set to "yes", the user receives
    ; the actual new/old message counts, not just a yes/no
    ; as to whether they have messages.  this can be set on
    ; a per-peer basis as well
    ;
    ;mailboxdetail=yes
    ;
    ; If regcontext is specified, Asterisk will dynamically 
    ; create and destroy a NoOp priority 1 extension for a given
    ; peer who registers or unregisters with us.  The actual extension
    ; is the 'regexten' parameter of the registering peer or its
    ; name if 'regexten' is not provided.  More than one regexten may be supplied
    ; if they are separated by '&'.  Patterns may be used in regexten.
    ;
    ;regcontext=iaxregistrations
    ;
    ; Guest sections for unauthenticated connection attempts.  Just
    ; specify an empty secret, or provide no secret section.
    ;
    [guest]
    type=user
    context=default
    callerid="Guest IAX User"
    
    ;
    ; Trust Caller*ID Coming from iaxtel.com
    ;
    [iaxtel]
    type=user
    context=default
    auth=rsa
    inkeys=iaxtel
    
    ;
    ; Trust Caller*ID Coming from iax.fwdnet.net
    ;
    [iaxfwd]
    type=user
    context=default
    auth=rsa
    inkeys=freeworlddialup
    
    ;
    ; Further user sections may be added, specifying a context and a
    ; secret used for connections with that given authentication name.
    ; Limited IP based access control is allowed by use of "allow" and
    ; "deny" keywords.  Multiple rules are permitted.  Multiple permitted
    ; contexts may be specified, in which case the first will be the default.
    ; You can also override caller*ID so that when you receive a call you
    ; set the Caller*ID to be what you want instead of trusting what
    ; the remote user provides
    ;
    ; There are three authentication methods that are supported:  md5, plaintext,
    ; and rsa.  The least secure is "plaintext", which sends passwords cleartext
    ; across the net.  "md5" uses a challenge/response md5 sum arrangement, but
    ; still requires both ends have plain text access to the secret.  "rsa" allows
    ; unidirectional secret knowledge through public/private keys.  If "rsa"
    ; authentication is used, "inkeys" is a list of acceptable public keys on the 
    ; local system that can be used to authenticate the remote peer, separated by
    ; the ":" character.  "outkey" is a single, private key to use to authenticate
    ; to the other side.  Public keys are named /var/lib/asterisk/keys/<name>.pub
    ; while private keys are named /var/lib/asterisk/keys/<name>.key.  Private
    ; keys should always be 3DES encrypted.
    ;
    ;
    ;[markster]
    ;type=user
    ;context=default
    ;context=local
    ;auth=md5,plaintext,rsa
    ;secret=markpasswd
    ;notransfer=yes		; Disable IAX native transfer
    ;jitterbuffer=yes	; Override global setting an enable jitter buffer
    ;			; for this user
    ;callerid="Mark Spencer" <(256) 428-6275>
    ;deny=0.0.0.0/0.0.0.0
    ;accountcode=markster0101
    ;permit=209.16.236.73/255.255.255.0
    ;language=en		; Use english as default language
    ;
    ; Peers may also be specified, with a secret and
    ; a remote hostname.
    ;
    [demo]
    type=peer
    username=asterisk
    secret=supersecret
    host=216.207.245.47
    ;sendani=no
    ;host=asterisk.linux-support.net
    ;port=5036
    ;mask=255.255.255.255
    ;qualify=yes	; Make sure this peer is alive
    ;jitterbuffer=no	; Turn off jitter buffer for this peer
    
    ;
    ; Peers can remotely register as well, so that they can be
    ; mobile.  Default IP's can also optionally be given but
    ; are not required.  Caller*ID can be suggested to the other
    ; side as well if it is for example a phone instead of another
    ; PBX.
    ;
    
    ;[dynamichost]
    ;host=dynamic
    ;secret=mysecret
    ;mailbox=1234		; Notify about mailbox 1234
    ;inkeys=key1:key2
    ;peercontext=local	; Default context to request for calls to peer
    ;defaultip=216.207.245.34
    ;callerid="Some Host" <(256) 428-6011>
    ;
    
    ;
    ;[biggateway]
    ;type=peer
    ;host=192.168.0.1
    ;context=*
    ;secret=myscret
    ;trunk=yes		; Use IAX2 trunking with this host
    ;
    
    ;
    ; Friends are a short cut for creating a user and
    ; a peer with the same values.
    ;
    ;[marko]
    ;type=friend
    ;host=dynamic
    ;regexten=1234
    ;secret=moofoo
    ;context=default
    ;permit=0.0.0.0/0.0.0.0
    
    [christopherhuebel]
    type=friend
    username=christopherhuebel
    host=dynamic
    secret=anything
    context=default
    
    [michaelweis]
    type=friend
    username=michaelweis
    host=dynamic
    secret=anything
    context=default
    extensions.conf
    Code:
    ;
    ; Static extension configuration file, used by
    ; the pbx_config module. This is where you configure all your 
    ; inbound and outbound calls in Asterisk. 
    ; 
    
    ;
    ; 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
    
    ; 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 ${VARIABLE} or ${ENV(VARIABLE)} for Environmental variable
    ; ${${VARIABLE}} or ${text${VARIABLE}} or any hybrid
    ;
    [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
    TRUNKMSD=1					; MSD digits to strip (usually 1 or 0)
    ;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)
    ;
    ; For example the extension _NXXXXXX would match normal 7 digit dialings, 
    ; while _1NXXNXXXXXX would represent an area code plus phone number
    ; preceeded by a one.
    ;
    ; Each step of an extension is ordered by priority, which must
    ; always start with 1 to be considered a valid extension.
    ;
    ; 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,application(arg1,arg2,...)
    ;exten => someexten,priority,application,arg1|arg2...
    ;
    ; Timing list for includes is 
    ;
    ;   <time range>|<days of week>|<days of month>|<months>
    ;
    ;include => daytime|9:00-17:00|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.
    ;
    
    ;
    ; 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 [url]www.gnophone.com[/url] or [url]www.iaxtel.com[/url]
    ;
    [iaxtel700]
    exten => _91700XXXXXXX,1,Dial(IAX2/${IAXINFO}@iaxtel.com/${EXTEN:1}@iaxtel)
    
    ;
    ; The SWITCH statement permits a server to share the dialplain 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,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
    exten => _9011.,2,Congestion
    
    [trunkld]
    ;
    ; Long distance context accessed through trunk
    ;
    exten => _91NXXNXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
    exten => _91NXXNXXXXXX,2,Congestion
    
    [trunklocal]
    ;
    ; Local seven-digit dialing accessed through trunk interface
    ;
    exten => _9NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
    exten => _9NXXXXXX,2,Congestion
    
    [trunktollfree]
    ;
    ; Long distance context accessed through trunk interface
    ;
    exten => _91800NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
    exten => _91800NXXXXXX,2,Congestion
    exten => _91888NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
    exten => _91888NXXXXXX,2,Congestion
    exten => _91877NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
    exten => _91877NXXXXXX,2,Congestion
    exten => _91866NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
    exten => _91866NXXXXXX,2,Congestion
    
    [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 => parkedcalls
    include => trunklocal
    include => iaxtel700
    include => trunktollfree
    include => iaxprovider
    ;
    ; 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
    
    [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(u${ARG1})		; 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(b${ARG1})			; 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
    
    [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,2,Answer			; Answer the line
    exten => s,3,DigitTimeout,5		; Set Digit Timeout to 5 seconds
    exten => s,4,ResponseTimeout,10		; Set Response Timeout to 10 seconds
    exten => s,5,BackGround(demo-congrats)	; Play a congratulatory message
    exten => s,6,BackGround(demo-instruct)	; Play some instructions
    
    exten => 2,1,BackGround(demo-moreinfo)	; Give some more information.
    exten => 2,2,Goto(s,6)
    
    exten => 3,1,SetLanguage(fr)		; Set language to french
    exten => 3,2,Goto(s,5)			; 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 => 1234,2,Macro(stdexten,1234,${CONSOLE})
    
    exten => 1235,1,Voicemail(u1234)		; Right to voicemail
    
    exten => 1236,1,Dial(Console/dsp)		; Ring forever
    exten => 1236,2,Voicemail(u1234)		; Unless busy
    
    ;
    ; # for when they're done with the demo
    ;
    exten => #,1,Playback(demo-thanks)		; "Thanks for trying the demo"
    exten => #,2,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 => 500,2,Dial(IAX2/guest@misery.digium.com/s@default)	; Call the Asterisk demo
    exten => 500,3,Playback(demo-nogo)	; Couldn't connect to the demo site
    exten => 500,4,Goto(s,6)		; Return to the start over message.
    
    ;
    ; Create an extension, 600, for evaulating echo latency.
    ;
    exten => 600,1,Playback(demo-echotest)	; Let them know what's going on
    exten => 600,2,Echo			; Do the echo test
    exten => 600,3,Playback(demo-echodone)	; Let them know it's over
    exten => 600,4,Goto(s,6)		; Start over
    
    ;
    ; Give voicemail at extension 8500
    ;
    exten => 8500,1,VoicemailMain
    exten => 8500,2,Goto(s,6)
    ;
    ; Here's what a phone entry would look like (IXJ for example)
    ;
    ;exten => 1265,1,Dial(Phone/phone0,15)
    ;exten => 1265,2,Goto(s,5)
    
    ;[mainmenu]
    ;
    ; Example "main menu" context with submenu
    ;
    ;exten => s,1,Answer
    ;exten => s,2,Background(thanks)		; "Thanks for calling press 1 for sales, 2 for support, ..."
    ;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,2,Wait,2
    ;exten => s,3,Background(submenuopts)	; "Thanks for calling the sales department.  Press 1 for steve, 2 for..."
    ;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
    
    ;
    ; Extensions like the two below can be used for FWD, Nikotel, sipgate etc.
    ; Note that you must have a [sipprovider] section in sip.conf whereas
    ; the otherprovider.net example does not require such a peer definition
    ;
    ;exten => _41X.,1,Dial(SIP/${EXTEN:2}@sipprovider,,r)
    ;exten => _42X.,1,Dial(SIP/user:passwd@${EXTEN:2}@otherprovider.net,30,rT)
    
    ; 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 ; Channel hints for presence
    ;exten => 6245,1,Dial(SIP/Grandstream1,20,rt)	; permit transfer
    ;exten => 6245,1,Dial(${HINT},20,rtT)		; Use hint as listed
    ;exten => 6361,1,Dial(IAX2/JaneDoe,,rm)		; ring without time limit
    ;exten => 6389,1,Dial(MGCP/aaln/1@192.168.0.14)
    ;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)
    ;
    ; Some other handy things are an extension for checking voicemail via
    ; voicemailmain
    ;
    ;exten => 8500,1,VoicemailMain
    ;exten => 8500,2,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))
    ;
    ; 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. 
    ;
    
    exten => 6666,1,Dial(IAX2/christopherhuebel)
    exten => 7777,1,Dial(IAX2/michaelweis)
    
    exten => s,1,Dial(CAPI/123456789:b${EXTEN},30)
    
    exten => _X.,1,Answer
    exten => _X.,2,Dial(CAPI/908475:${EXTEN},,Ttr)
    exten => _X.,3,Congestion
    exten => _X.,4,Hangup
    capi.conf
    Code:
    ;
    ; CAPI config
    ;
    ;
    [general]
    nationalprefix=0
    internationalprefix=00
    rxgain=0.8
    txgain=0.8
    
    [interfaces]
    
    msn=908475
    incomingmsn=*
    controller=1
    softdtmf=1
    accountcode=
    context=demo
    ;echosquelch=1
    ;echocancel=yes
    ;echotail=64
    ;callgroup=1
    deflect=123456789
    devices=2
    
    
    ;PointToPoint (55512-0)
    ;for outgoing calls use example 5551212
    ;and in dialplan you can use callerid like
    ;exten => _0XXX.,1,StripMSD,1
    ;exten => _XXX.,2,Dial,CAPI/55512${CALLERIDNUM}:bBYEXTENSION
    ;============================================================
    ;mode=immediate
    ;isdnmode=ptp
    ;msn=55512
    ;controller=2
    ;devices=30
    
    exten => _0XXX.,1,StripMSD,1
    exten => _XXX.,2,Dial,CAPI/908475${CALLERIDNUM}:bBYEXTENSION 
     
  2. kombjuder

    kombjuder IPPF-Promi

    Registriert seit:
    2 Nov. 2004
    Beiträge:
    3,086
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    0
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    Ort:
    Weil am Rhein
    Re: Konfiguration

    Die Firewalleinstellungen sind vermutlich fehlerhaft. Den Asterisk mal in die DMZ stellen. Wenn's dann geht, die entsprechenden Ports auf dem Router freigeben.
     
  3. topher

    topher Neuer User

    Registriert seit:
    17 Aug. 2005
    Beiträge:
    19
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    0
    Punkte für Erfolge:
    0
    Die Firewalls sind an meinem PC sowie am Linux-PC abgeschaltet..

    Welche Ports müsste man denn freigeben?

    Das merkwürdige ist ja, dass Ton zwar rausgeht, aber nicht reinkommt. Am Telefon kann's auch nicht liegen. Wir haben das schon mit verschiedenen Telefonen getestet.

    Hat das vielleicht was mit dem "chan_skinny" zu tun?

    Hier noch mein Log:
    Code:
      == Parsing '/etc/asterisk/asterisk.conf': Found
      == Parsing '/etc/asterisk/extconfig.conf': Found
    Asterisk 1.0.6-BRIstuffed-0.2.0-RC7k, Copyright (C) 1999-2004 Digium.
    Written by Mark Spencer <markster@digium.com>
    =========================================================================
      == Parsing '/etc/asterisk/logger.conf': Found
    Asterisk Event Logger Started /var/log/asterisk/event_log
      == Manager registered action Ping
      == Manager registered action Events
      == Manager registered action Logoff
      == Manager registered action Hangup
      == Manager registered action Status
      == Manager registered action Setvar
      == Manager registered action Getvar
      == Manager registered action Redirect
      == Manager registered action Originate
      == Manager registered action Command
      == Manager registered action ExtensionState
      == Manager registered action AbsoluteTimeout
      == Manager registered action MailboxStatus
      == Manager registered action MailboxCount
      == Manager registered action DBget
      == Manager registered action DBput
      == Manager registered action DBdel
      == Manager registered action ListCommands
      == Parsing '/etc/asterisk/manager.conf': Found
      == Parsing '/etc/asterisk/rtp.conf': Found
      == RTP Allocating from port range 10000 -> 20000
    Asterisk PBX Core Initializing
    Registering builtin applications:
     [AbsoluteTimeout]
      == Registered application 'AbsoluteTimeout'
     [Answer]
      == Registered application 'Answer'
     [BackGround]
      == Registered application 'BackGround'
     [Busy]
      == Registered application 'Busy'
     [Congestion]
      == Registered application 'Congestion'
     [DigitTimeout]
      == Registered application 'DigitTimeout'
     [Goto]
      == Registered application 'Goto'
     [GotoIf]
      == Registered application 'GotoIf'
     [GotoIfTime]
      == Registered application 'GotoIfTime'
     [Hangup]
      == Registered application 'Hangup'
     [NoOp]
      == Registered application 'NoOp'
     [Prefix]
      == Registered application 'Prefix'
     [Progress]
      == Registered application 'Progress'
     [ResetCDR]
      == Registered application 'ResetCDR'
     [ResponseTimeout]
      == Registered application 'ResponseTimeout'
     [Ringing]
      == Registered application 'Ringing'
     [SayNumber]
      == Registered application 'SayNumber'
     [SayDigits]
      == Registered application 'SayDigits'
     [SayAlpha]
      == Registered application 'SayAlpha'
     [SayPhonetic]
      == Registered application 'SayPhonetic'
     [SetAccount]
      == Registered application 'SetAccount'
     [SetAMAFlags]
      == Registered application 'SetAMAFlags'
     [SetGlobalVar]
      == Registered application 'SetGlobalVar'
     [SetLanguage]
      == Registered application 'SetLanguage'
     [SetVar]
      == Registered application 'SetVar'
     [StripMSD]
      == Registered application 'StripMSD'
     [Suffix]
      == Registered application 'Suffix'
     [Wait]
      == Registered application 'Wait'
     [WaitExten]
      == Registered application 'WaitExten'
    Asterisk Dynamic Loader Starting:
      == Parsing '/etc/asterisk/modules.conf': Found
     [chan_capi.so] => (Common ISDN API for Asterisk)
      == Parsing '/etc/asterisk/capi.conf': Found
        -- This box has 1 capi controller(s).
        -- CAPI[contr1] supports DTMF
        -- CAPI[contr1] supports supplementary services
           > HOLD/RETRIEVE
           > TERMINAL PORTABILITY
           > ECT
           > 3PTY
           > CF
           > CD
           > MCID
           > CCBS
           > MWI
           > CCNR
      == ast_capi_pvt(908475,908475,default,0,2) (1,2,64)
      == ast_capi_pvt(908475,908475,default,0,2) (1,2,64)
        -- listening on contr1 CIPmask = 0x1fff03ff
      == Registered channel type 'CAPI' (Common ISDN API Driver (0.3.5) aLaw CVS HEAD)
     [res_features.so] => (Call Parking Resource)
      == Parsing '/etc/asterisk/features.conf': Found
        -- Registered extension context 'parkedcalls'
        -- Added extension '700' priority 1 to parkedcalls
      == Registered application 'ParkedCall'
      == Registered application 'Park'
      == Manager registered action ParkedCalls
      == Registered application 'HoldedCall'
      == Registered application 'AutoanswerLogin'
      == Registered application 'Autoanswer'
     [res_musiconhold.so] => (Music On Hold Resource)
      == Parsing '/etc/asterisk/musiconhold.conf': Found
    Aug 24 07:41:37 WARNING[7038]: res_musiconhold.c:565 moh_register: Unable to open pseudo channel for timing...  Sound may be choppy.
      == Registered application 'MusicOnHold'
      == Registered application 'WaitMusicOnHold'
      == Registered application 'SetMusicOnHold'
     [res_indications.so] => (Indications Configuration)
      == Parsing '/etc/asterisk/indications.conf': Found
        -- Registered indication country 'cl'
        -- Registered indication country 'tw'
        -- Registered indication country 'us'
        -- Registered indication country 'au'
        -- Registered indication country 'fr'
        -- Registered indication country 'de'
        -- Registered indication country 'nl'
        -- Registered indication country 'uk'
        -- Registered indication country 'fi'
        -- Registered indication country 'no'
        -- Registered indication country 'br'
        -- Registered indication country 'za'
        -- Registered indication country 'it'
        -- Registered indication country 'us-o'
        -- Registered indication country 'gr'
        -- Registered indication country 'ru'
        -- Registered indication country 'nz'
        -- Setting default indication country to 'us'
      == Registered application 'Playtones'
      == Registered application 'StopPlaytones'
     [res_agi.so] => (Asterisk Gateway Interface (AGI))
      == Registered application 'DeadAGI'
      == Registered application 'EAGI'
      == Registered application 'AGI'
     [res_crypto.so] => (Cryptographic Digital Signatures)
        -- Loaded PUBLIC key 'iaxtel'
        -- Loaded PUBLIC key 'freeworlddialup'
     [res_adsi.so] => (ADSI Resource)
      == Parsing '/etc/asterisk/adsi.conf': Found
     [res_monitor.so] => (Call Monitoring Resource)
      == Registered application 'Monitor'
      == Registered application 'StopMonitor'
      == Registered application 'ChangeMonitor'
      == Manager registered action Monitor
      == Manager registered action StopMonitor
      == Manager registered action ChangeMonitor
     [app_sms.so] => (SMS/PSTN handler)
      == Registered application 'SMS'
     [app_hasnewvoicemail.so] => (Indicator for whether a voice mailbox has messages in a given folder.
      == Registered application 'HasVoicemail'
      == Registered application 'HasNewVoicemail'
     [format_wav_gsm.so] => (Microsoft WAV format (Proprietary GSM))
      == Registered file format wav49, extension(s) WAV|wav49
     [app_url.so] => (Send URL Applications)
      == Registered application 'SendURL'
     [skipping chan_modem_i4l.so]
     [app_test.so] => (Interface Test Application)
      == Registered application 'TestClient'
      == Registered application 'TestServer'
     [chan_mgcp.so] => (Media Gateway Control Protocol (MGCP))
      == Parsing '/etc/asterisk/mgcp.conf': Found
      == MGCP Listening on 0.0.0.0:2727
      == Using TOS bits 0
      == Registered channel type 'MGCP' (Media Gateway Control Protocol (MGCP))
     [app_eval.so] => (Reevaluates strings)
      == Registered application 'Eval'
     [chan_zap.so] => (Zapata Telephony w/PRI)
      == Parsing '/etc/asterisk/zapata.conf': Found
        -- Automatically generated pseudo channel
      == Registered channel type 'Zap' (Zapata Telephony Driver w/PRI)
      == Registered channel type 'Tor' (Zapata Telephony Driver w/PRI)
      == Registered application 'CallingPres'
      == Registered application 'zapEC'
      == Manager registered action ZapTransfer
      == Manager registered action ZapHangup
      == Manager registered action ZapDialOffhook
      == Manager registered action ZapDNDon
      == Manager registered action ZapDNDoff
      == Manager registered action ZapShowChannels
     [app_sendtext.so] => (Send Text Applications)
      == Registered application 'SendText'
     [app_exec.so] => (Executes applications)
      == Registered application 'Exec'
     [app_txtcidname.so] => (TXTCIDName)
      == Registered application 'TXTCIDName'
      == Parsing '/etc/asterisk/enum.conf': Found
     [cdr_manager.so] => (Asterisk Call Manager CDR Backend)
      == Parsing '/etc/asterisk/cdr_manager.conf': Found
     [app_capiCD.so] => ((CAPI*) Call Deflection, the magic thing.)
      == Registered application 'capiCD'
     [app_directory.so] => (Extension Directory)
      == Registered application 'Directory'
     [app_playback.so] => (Trivial Playback Application)
      == Registered application 'Playback'
     [app_capiNoES.so] => ((CAPI*) No Echo Suppression.)
      == Registered application 'capiNoES'
     [codec_adpcm.so] => (Adaptive Differential PCM Coder/Decoder)
      == Registered translator 'adpcmtolin' from format adpcm to slin, cost 1
      == Registered translator 'lintoadpcm' from format slin to adpcm, cost 1
     [chan_local.so] => (Local Proxy Channel)
      == Registered channel type 'Local' (Local Proxy Channel Driver)
     [app_groupcount.so] => (Group Management Routines)
      == Registered application 'GetGroupCount'
      == Registered application 'SetGroup'
      == Registered application 'CheckGroup'
     [app_adsiprog.so] => (Asterisk ADSI Programming Application)
      == Registered application 'ADSIProg'
     [app_chanisavail.so] => (Check if channel is available)
      == Registered application 'ChanIsAvail'
     [app_qcall.so] => (Call from Queue)
     [app_softhangup.so] => (Hangs up the requested channel)
      == Registered application 'SoftHangup'
     [codec_lpc10.so] => (LPC10 2.4kbps (signed linear) Voice Coder)
      == Registered translator 'lpc10tolin' from format lpc10 to slin, cost 6
      == Registered translator 'lintolpc10' from format slin to lpc10, cost 21
     [app_setcidname.so] => (Set CallerID Name)
      == Registered application 'SetCIDName'
     [format_g726.so] => (Raw G.726 (16/24/32/40kbps) data)
      == Registered file format g726-40, extension(s) g726-40
      == Registered file format g726-32, extension(s) g726-32
      == Registered file format g726-24, extension(s) g726-24
      == Registered file format g726-16, extension(s) g726-16
     [format_g729.so] => (Raw G729 data)
      == Registered file format g729, extension(s) g729
     [app_userevent.so] => (Custom User Event Application)
      == Registered application 'UserEvent'
     [codec_gsm.so] => (GSM/PCM16 (signed linear) Codec Translator)
      == Registered translator 'gsmtolin' from format gsm to slin, cost 2
      == Registered translator 'lintogsm' from format slin to gsm, cost 6
     [app_authenticate.so] => (Authentication Application)
      == Registered application 'Authenticate'
     [format_pcm_alaw.so] => (Raw aLaw 8khz PCM Audio support)
      == Registered file format alaw, extension(s) alaw|al
     [format_ilbc.so] => (Raw iLBC data)
      == Registered file format iLBC, extension(s) ilbc
     [format_h263.so] => (Raw h263 data)
      == Registered file format h263, extension(s) h263
     [app_forkcdr.so] => (Fork The CDR into 2 separate entities.)
      == Registered application 'ForkCDR'
     [app_ices.so] => (Encode and Stream via icecast and ices)
      == Registered application 'ICES'
     [app_nbscat.so] => (Silly NBS Stream Application)
      == Registered application 'NBScat'
     [codec_a_mu.so] => (A-law and Mulaw direct Coder/Decoder)
      == Registered translator 'alawtoulaw' from format alaw to ulaw, cost 1
      == Registered translator 'ulawtoalaw' from format ulaw to alaw, cost 1
     [app_system.so] => (Generic System() application)
      == Registered application 'TrySystem'
      == Registered application 'System'
     [app_record.so] => (Trivial Record Application)
      == Registered application 'Record'
     [chan_iax2.so] => (Inter Asterisk eXchange (Ver 2))
    Aug 24 07:41:37 WARNING[7038]: chan_iax2.c:7487 load_module: Unable to open IAX timing interface: No such file or directory
      == Manager registered action IAXpeers
      == Parsing '/etc/asterisk/iax.conf': Found
        -- Seeding 'christopherhuebel' at 149.249.226.106:4569 for 60
      == Registered channel type 'IAX2' (Inter Asterisk eXchange Driver (Ver 2))
      == Using TOS bits 16
      == IAX Ready and Listening on 0.0.0.0 port 4569
      == Loaded firmware 'iaxy.bin'
      == Parsing '/etc/asterisk/iaxprov.conf': Found
        -- Loaded provisioning template 'default'
     [app_milliwatt.so] => (Digital Milliwatt (mu-law) Test Application)
      == Registered application 'Milliwatt'
     [app_parkandannounce.so] => (Call Parking and Announce Application)
      == Registered application 'ParkAndAnnounce'
     [app_sayunixtime.so] => (Say time)
      == Registered application 'SayUnixTime'
      == Registered application 'DateTime'
     [pbx_spool.so] => (Outgoing Spool Support)
     [app_capiMCID.so] => ((CAPI*) Malicious Caller ID, the evil thing.)
      == Registered application 'capiMCID'
     [app_zapscan.so] => (Scan Zap channels application)
      == Registered application 'ZapScan'
     [app_macro.so] => (Extension Macros)
      == Registered application 'Macro'
     [app_random.so] => (Random goto)
      == Registered application 'Random'
     [codec_ulaw.so] => (Mu-law Coder/Decoder)
      == Registered translator 'ulawtolin' from format ulaw to slin, cost 1
      == Registered translator 'lintoulaw' from format slin to ulaw, cost 1
     [app_zapras.so] => (Zap RAS Application)
      == Registered application 'ZapRAS'
     [app_capiRETRIEVE.so] => ((CAPI*) RETRIEVE)
      == Registered application 'capiRETRIEVE'
     [chan_agent.so] => (Agent Proxy Channel)
      == Registered channel type 'Agent' (Call Agent Proxy Channel)
      == Registered application 'AgentLogin'
      == Registered application 'AgentCallbackLogin'
      == Registered application 'AgentMonitorOutgoing'
      == Parsing '/etc/asterisk/agents.conf': Found
     [app_controlplayback.so] => (Control Playback Application)
      == Registered application 'ControlPlayback'
     [format_jpeg.so] => (JPEG (Joint Picture Experts Group) Image Format)
      == Registered format 'jpg' (JPEG (Joint Picture Experts Group))
     [codec_alaw.so] => (A-law Coder/Decoder)
      == Registered translator 'alawtolin' from format alaw to slin, cost 1
      == Registered translator 'lintoalaw' from format slin to alaw, cost 1
     [app_transfer.so] => (Transfer)
      == Registered application 'Transfer'
     [cdr_csv.so] => (Comma Separated Values CDR Backend)
     [app_voicemail.so] => (Comedian Mail (Voicemail System))
      == Registered application 'VoiceMail'
      == Registered application 'VoiceMail2'
      == Registered application 'VoiceMailMain'
      == Registered application 'VoiceMailMain2'
      == Registered application 'MailboxExists'
      == Parsing '/etc/asterisk/voicemail.conf': Found
     [app_pickup.so] => (PickUp/PickDown/Steal/PickupChan)
      == Registered application 'PickupChan'
      == Registered application 'PickDown'
      == Registered application 'Steal'
      == Registered application 'PickUp'
     [codec_speex.so] => (Speex/PCM16 (signed linear) Codec Translator)
      == Registered translator 'speextolin' from format speex to slin, cost 4
      == Registered translator 'lintospeex' from format slin to speex, cost 95
     [app_verbose.so] => (Send verbose output)
      == Registered application 'Verbose'
     [app_setcdruserfield.so] => (CDR user field apps)
      == Registered application 'SetCDRUserField'
      == Registered application 'AppendCDRUserField'
      == Manager registered action SetCDRUserField
     [codec_g726.so] => (ITU G.726-32kbps G726 Transcoder)
      == Registered translator 'g726tolin' from format g726 to slin, cost 8
      == Registered translator 'lintog726' from format slin to g726, cost 9
     [app_lookupblacklist.so] => (Look up Caller*ID name/number from blacklist database)
      == Registered application 'LookupBlacklist'
     [app_zapbarge.so] => (Barge in on Zap channel application)
      == Registered application 'ZapBarge'
     [app_getcpeid.so] => (Get ADSI CPE ID)
      == Registered application 'GetCPEID'
     [app_enumlookup.so] => (ENUM Lookup)
      == Registered application 'EnumLookup'
      == Parsing '/etc/asterisk/enum.conf': Found
     [codec_ilbc.so] => (iLBC/PCM16 (signed linear) Codec Translator)
      == Registered translator 'ilbctolin' from format ilbc to slin, cost 25
      == Registered translator 'lintoilbc' from format slin to ilbc, cost 51
     [pbx_config.so] => (Text Extension Configuration)
      == Parsing '/etc/asterisk/extensions.conf': Found
        -- Setting global variable 'CONSOLE' to 'Console/dsp'
        -- Setting global variable 'IAXINFO' to 'guest'
        -- Setting global variable 'TRUNK' to 'Zap/g2'
        -- Setting global variable 'TRUNKMSD' to '1'
        -- Registered extension context 'iaxtel700'
        -- Added extension '_91700XXXXXXX' priority 1 to iaxtel700
        -- Registered extension context 'iaxprovider'
        -- Registered extension context 'trunkint'
        -- Added extension '_9011.' priority 1 to trunkint
        -- Added extension '_9011.' priority 2 to trunkint
        -- Registered extension context 'trunkld'
        -- Added extension '_91NXXNXXXXXX' priority 1 to trunkld
        -- Added extension '_91NXXNXXXXXX' priority 2 to trunkld
        -- Registered extension context 'trunklocal'
        -- Added extension '_9NXXXXXX' priority 1 to trunklocal
        -- Added extension '_9NXXXXXX' priority 2 to trunklocal
        -- Registered extension context 'trunktollfree'
        -- Added extension '_91800NXXXXXX' priority 1 to trunktollfree
        -- Added extension '_91800NXXXXXX' priority 2 to trunktollfree
        -- Added extension '_91888NXXXXXX' priority 1 to trunktollfree
        -- Added extension '_91888NXXXXXX' priority 2 to trunktollfree
        -- Added extension '_91877NXXXXXX' priority 1 to trunktollfree
        -- Added extension '_91877NXXXXXX' priority 2 to trunktollfree
        -- Added extension '_91866NXXXXXX' priority 1 to trunktollfree
        -- Added extension '_91866NXXXXXX' priority 2 to trunktollfree
        -- Registered extension context 'international'
        -- Including context 'longdistance' in context 'international'
        -- Including context 'trunkint' in context 'international'
        -- Registered extension context 'longdistance'
        -- Including context 'local' in context 'longdistance'
        -- Including context 'trunkld' in context 'longdistance'
        -- Registered extension context 'local'
        -- Including context 'default' in context 'local'
        -- Including context 'parkedcalls' in context 'local'
        -- Including context 'trunklocal' in context 'local'
        -- Including context 'iaxtel700' in context 'local'
        -- Including context 'trunktollfree' in context 'local'
        -- Including context 'iaxprovider' in context 'local'
        -- Registered extension context 'macro-stdexten'
        -- Added extension 's' priority 1 to macro-stdexten
        -- Added extension 's' priority 2 to macro-stdexten
        -- Added extension 's-NOANSWER' priority 1 to macro-stdexten
        -- Added extension 's-NOANSWER' priority 2 to macro-stdexten
        -- Added extension 's-BUSY' priority 1 to macro-stdexten
        -- Added extension 's-BUSY' priority 2 to macro-stdexten
        -- Added extension '_s-.' priority 1 to macro-stdexten
        -- Added extension 'a' priority 1 to macro-stdexten
        -- Registered extension context 'demo'
        -- Added extension 's' priority 1 to demo
        -- Added extension 's' priority 2 to demo
        -- Added extension 's' priority 3 to demo
        -- Added extension 's' priority 4 to demo
        -- Added extension 's' priority 5 to demo
        -- Added extension 's' priority 6 to demo
        -- Added extension '2' priority 1 to demo
        -- Added extension '2' priority 2 to demo
        -- Added extension '3' priority 1 to demo
        -- Added extension '3' priority 2 to demo
        -- Added extension '1000' priority 1 to demo
        -- Added extension '1234' priority 1 to demo
        -- Added extension '1234' priority 2 to demo
        -- Added extension '1235' priority 1 to demo
        -- Added extension '1236' priority 1 to demo
        -- Added extension '1236' priority 2 to demo
        -- Added extension '#' priority 1 to demo
        -- Added extension '#' priority 2 to demo
        -- Added extension 't' priority 1 to demo
        -- Added extension 'i' priority 1 to demo
        -- Added extension '500' priority 1 to demo
        -- Added extension '500' priority 2 to demo
        -- Added extension '500' priority 3 to demo
        -- Added extension '500' priority 4 to demo
        -- Added extension '600' priority 1 to demo
        -- Added extension '600' priority 2 to demo
        -- Added extension '600' priority 3 to demo
        -- Added extension '600' priority 4 to demo
        -- Added extension '8500' priority 1 to demo
        -- Added extension '8500' priority 2 to demo
        -- Registered extension context 'default'
        -- Including context 'demo' in context 'default'
        -- Added extension '6666' priority 1 to default
        -- Added extension '7777' priority 1 to default
        -- Added extension 's' priority 1 to default
        -- Added extension '_X.' priority 1 to default
        -- Added extension '_X.' priority 2 to default
        -- Added extension '_X.' priority 3 to default
        -- Added extension '_X.' priority 4 to default
     [app_segfault.so] => (Application for crashing Asterisk with a segmentation fault)
      == Registered application 'Segfault'
     [app_read.so] => (Read Variable Application)
      == Registered application 'Read'
     [app_alarmreceiver.so] => (Alarm Receiver for Asterisk)
      == Parsing '/etc/asterisk/alarmreceiver.conf': Found
      == Registered application 'AlarmReceiver'
     [format_gsm.so] => (Raw GSM data)
      == Registered file format gsm, extension(s) gsm
     [app_dial.so] => (Dialing Application)
      == Registered application 'Dial'
     [app_striplsd.so] => (Strip trailing digits)
      == Registered application 'StripLSD'
     [app_capiECT.so] => ((CAPI*) ECT)
      == Registered application 'capiECT'
     [app_disa.so] => (DISA (Direct Inward System Access) Application)
      == Registered application 'DISA'
     [app_cdr.so] => (Make sure asterisk doesn't save CDR for a certain call)
      == Registered application 'NoCDR'
     [app_image.so] => (Image Transmission Application)
      == Registered application 'SendImage'
     [skipping chan_modem_bestdata.so]
     [app_cut.so] => (Cuts up variables)
      == Registered application 'Cut'
     [app_devstate.so] => (Application for sending device state messages)
      == Registered channel type 'DS' (Application for sending device state messages)
      == Manager registered action Devstate
      == Registered application 'Devstate'
     [skipping chan_modem.so]
     [app_festival.so] => (Simple Festival Interface)
      == Registered application 'Festival'
     [app_meetme.so] => (MeetMe conference bridge)
      == Registered application 'MeetMeAdmin'
      == Registered application 'MeetMeCount'
      == Registered application 'MeetMe'
     [app_echo.so] => (Simple Echo Application)
      == Registered application 'Echo'
     [chan_phone.so] => (Linux Telephony API Support)
      == Parsing '/etc/asterisk/phone.conf': Found
      == Registered channel type 'Phone' (Standard Linux Telephony API Driver)
     [format_pcm.so] => (Raw uLaw 8khz Audio support (PCM))
      == Registered file format pcm, extension(s) pcm|ulaw|ul|mu
     [app_privacy.so] => (Require phone number to be entered, if no CallerID sent)
      == Registered application 'PrivacyManager'
     [app_flash.so] => (Flash zap trunk application)
      == Registered application 'Flash'
     [skipping app_intercom.so]
     [app_setcallerid.so] => (Set CallerID Application)
      == Registered application 'SetCallerPres'
      == Registered application 'SetCallerID'
     [pbx_wilcalu.so] => (Wil Cal U (Auto Dialer))
     [app_capiHOLD.so] => ((CAPI*) HOLD)
      == Registered application 'capiHOLD'
     [app_substring.so] => ((Deprecated) Save substring digits in a given variable)
      == Registered application 'SubString'
     [chan_skinny.so] => (Skinny Client Control Protocol (Skinny))
      == Parsing '/etc/asterisk/skinny.conf': Found
    Aug 24 07:41:37 WARNING[7038]: chan_skinny.c:2584 reload_config: Unable to get our IP address, Skinny disabled
      == Registered channel type 'Skinny' (Skinny Client Control Protocol (Skinny))
     [format_sln.so] => (Raw Signed Linear Audio support (SLN))
      == Registered file format sln, extension(s) sln|raw
     [app_zapateller.so] => (Block Telemarketers with Special Information Tone)
      == Registered application 'Zapateller'
     [app_queue.so] => (True Call Queueing)
      == Registered application 'Queue'
      == Manager registered action Queues
      == Manager registered action QueueStatus
      == Manager registered action QueueAdd
      == Manager registered action QueueRemove
      == Registered application 'AddQueueMember'
      == Registered application 'RemoveQueueMember'
      == Parsing '/etc/asterisk/queues.conf': Found
     [app_mp3.so] => (Silly MP3 Application)
      == Registered application 'MP3Player'
     [app_lookupcidname.so] => (Look up CallerID Name from local database)
      == Registered application 'LookupCIDName'
     [format_wav.so] => (Microsoft WAV format (8000hz Signed Linear))
      == Registered file format wav, extension(s) wav
     [app_senddtmf.so] => (Send DTMF digits Application)
      == Registered application 'SendDTMF'
     [format_vox.so] => (Dialogic VOX (ADPCM) File Format)
      == Registered file format vox, extension(s) vox
     [skipping chan_modem_aopen.so]
     [app_waitforring.so] => (Waits until first ring after time)
      == Registered application 'WaitForRing'
     [app_setcidnum.so] => (Set CallerID Number)
      == Registered application 'SetCIDNum'
     [chan_oss.so] => (OSS Console Channel Driver)
      == Console is full duplex
      == Registered channel type 'Console' (OSS Console Channel Driver)
      == Parsing '/etc/asterisk/oss.conf': Found
     [app_talkdetect.so] => (Playback with Talk Detection)
      == Registered application 'BackgroundDetect'
     [app_db.so]Aug 24 07:41:37 WARNING[7038]: chan_oss.c:239 sound_thread: Read error on sound device: Resource temporarily unavailable
     => (Database access functions for Asterisk extension logic)
      == Registered application 'DBget'
      == Registered application 'DBput'
      == Registered application 'DBdel'
      == Registered application 'DBdeltree'
     [chan_sip.so] => (Session Initiation Protocol (SIP))
      == Parsing '/etc/asterisk/sip.conf': Found
      == SIP Listening on 0.0.0.0:5060
      == Using TOS bits 0
      == Registered channel type 'SIP' (Session Initiation Protocol (SIP))
      == Registered application 'SIPDtmfMode'
      == Parsing '/etc/asterisk/enum.conf': Found
      == Parsing '/etc/asterisk/extconfig.conf': Found
      == Parsing '/etc/asterisk/logger.conf': Found
    Asterisk Event Logger restarted
      == Parsing '/etc/asterisk/manager.conf': Found
      == Parsing '/etc/asterisk/enum.conf': Found
      == Parsing '/etc/asterisk/rtp.conf': Found
      == RTP Allocating from port range 10000 -> 20000
    Asterisk Ready.
    
     
  4. Fell

    Fell Neuer User

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    Frage welches OS und welchen Kernel hast du.?
    Desweiteren verwendest du chan_capi oder chan_capi_cm?
     
  5. topher

    topher Neuer User

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    Was genau meinst du mit OS? Hab die Angaben soweit mal in meine Signatur geschrieben..
     
  6. Fell

    Fell Neuer User

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  7. armincm

    armincm Aktives Mitglied

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    chan_capi-0.3.5 hat definitiv ein problem unter kernels >=2.6.10. Es sollte auf jeden Fall ein neueres chan_capi verwendet werden -> sourceforge.

    Armin
     
  8. topher

    topher Neuer User

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    Dank dir! Es lag wirklich am chan_capi. Hab den ganzen Spaß nun neuinstalliert und habe nun asterisk 1.0.9 und chan_capi_cm. Damit funktioniert es einwandfrei auf der SuSE 9.3 : )