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Problem

Dieses Thema im Forum "Asterisk Skripte" wurde erstellt von Bintangku, 18 Apr. 2005.

  1. Bintangku

    Bintangku Neuer User

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    also ich habe einen asterisk installiert, hab es so konfiguriert das ich intern vom netzwerk telefonieren kann, aber wenn ich jetzt z.b meine sipgate telefonnr von einem dieser telefone anrufen will klappt es nicht es kommt immer der Fehler:

    chan.sip.c:2773 process_sdp: No compatible codec!
    pbx.c:1330 pbx_extension_helper: Cannot find extension contex "

    ich weiß nicht was ich machen soll!!

    ich habe meinen sipgate user auch in der sip.conf und in der extension.conf eingetragen, und wennich in der CLI sip show registry oder sip show users eingebe , zeigt esm mir auch an das der status dieses sipgate accounts ok sein. wenn ich jedoch auf meinen account von sipgate gehe, steht da nicht online, obwohl ja jetzt dieser account über asterisk laufen sollte, oder??

    ich bin ein total Neuling auf diesem Gebiet, kann mir bitt jemand helfen????????

    danke
     
  2. rajo

    rajo Admin-Team

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    hallo & willkommen,

    zeig doch mal Deine sip.conf und extensions.conf her.

    Bei dem chan.sip-fehler hast Du dem * ein paar audiocodecs zuviel verboten. allow=ulaw etc. sollte da schonmal Abhilfe schaffen.

    Das zweite schaut nach Tippfehler aus.
     
  3. Bintangku

    Bintangku Neuer User

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    also meine sip config:
    Code:
    ;
    ; SIP Configuration for Asterisk
    ;
    ; Syntax for specifying a SIP device in extensions.conf is
    ; SIP/devicename where devicename is defined in a section below.
    ;
    ; You may also use 
    ; SIP/username@domain to call any SIP user on the Internet
    ; (Don't forget to enable DNS SRV records if you want to use this)
    ; 
    ; If you define a SIP proxy as a peer below, you may call
    ; SIP/proxyhostname/user or SIP/user@proxyhostname 
    ; where the proxyhostname is defined in a section below 
    ; 
    ; Useful CLI commands to check peers/users:
    ;   sip show peers		Show all SIP peers (including friends)
    ;   sip show users		Show all SIP users (including friends)
    ;   sip show registry		Show status of hosts we register with
    ;
    ;   sip debug			Show all SIP messages
    ;
    
    [general]
    context=		; Default context for incoming calls
    ;recordhistory=yes		; Record SIP history by default 
    				; (see sip history / sip no history)
    ;realm=mydomain.tld		; Realm for digest authentication
    				; defaults to "asterisk"
    				; Realms MUST be globally unique according to RFC 3261
    				; Set this to your host name or domain name
    port=5060			; UDP Port to bind to (SIP standard port is 5060)
    bindaddr=0.0.0.0		; IP address to bind to (0.0.0.0 binds to all)
    srvlookup=yes			; Enable DNS SRV lookups on outbound calls
    				; Note: Asterisk only uses the first host 
    				; in SRV records
    				; Disabling DNS SRV lookups disables the 
    				; ability to place SIP calls based on domain 
    				; names to some other SIP users on the Internet
    				
    ;pedantic=yes			; Enable slow, pedantic checking for Pingtel
    				; and multiline formatted headers for strict
    				; SIP compatibility (defaults to "no")
    tos=0x18                        ; Set IP QoS to either a keyword or numeric val
    ;tos=lowdelay                   ; lowdelay,throughput,reliability,mincost,none
    ;maxexpirey=3600		; Max length of incoming registration we allow
    ;defaultexpirey=120		; Default length of incoming/outoing registration
    ;notifymimetype=text/plain	; Allow overriding of mime type in MWI NOTIFY
    ;videosupport=yes		; Turn on support for SIP video
    
    disallow=all			; First disallow all codecs
    allow=gsm			; Allow codecs in order of preference
    ;allow=ilbc			; Note: codec order is respected only in [general]
    ;musicclass=default		; Sets the default music on hold class for all SIP calls
    				; This may also be set for individual users/peers
    language=de			; Default language setting for all users/peers
    				; This may also be set for individual users/peers
    ;relaxdtmf=yes			; Relax dtmf handling
    ;rtptimeout=60			; Terminate call if 60 seconds of no RTP activity
    				; when we're not on hold
    ;rtpholdtimeout=300		; Terminate call if 300 seconds of no RTP activity
    				; when we're on hold (must be > rtptimeout)
    ;trustrpid = no			; If Remote-Party-ID should be trusted
    ;progressinband=no		; If we should generate in-band ringing always
    ;useragent=Asterisk PBX		; Allows you to change the user agent string
    nat=no				; NAT settings 
                                    ; yes = Always ignore info and assume NAT
                                    ; no = Use NAT mode only according to RFC3581 
                                    ; never = Never attempt NAT mode or RFC3581 support
    				; route = Assume NAT, don't send rport (work around more UNIDEN bugs)
    ;promiscredir = no      ; If yes, allows 302 or REDIR to non-local SIP address
    ;                       ; Note that promiscredir when redirects are made to the
    ;                       ; local system will cause loops since SIP is incapable
    ;                       ; of performing a "hairpin" call.
    ;
    ; 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
    ;
    ; Asterisk can register as a SIP user agent to a SIP proxy (provider)
    ; Format for the register statement is:
    ;       register => user[:secret[:authuser]]@host[:port][/extension]
    ;
    ; If no extension is given, the 's' extension is used. The extension
    ; needs to be defined in extensions.conf to be able to accept calls
    ; from this SIP proxy (provider)
    ;
    ; host is either a host name defined in DNS or the name of a 
    ; section defined below.
    ;
    ; Examples:
    ;
    ;register => 1234:password@mysipprovider.com	
    ;
    ;     This will pass incoming calls to the 's' extension
    ;
    ;
    register => 5559972:*****@sipgate.de/5559972
    ;
    ;    Register 2345 at sip provider 'sip_proxy'.  Calls from this provider connect to local 
    ;    extension 1234 in extensions.conf default context, unless you define 
    ;    unless you configure a [sip_proxy] section below, and configure a context.
    ;	 Tip 1: Avoid assigning hostname to a sip.conf section like [provider.com]
    ;        Tip 2: Use separate type=peer and type=user sections for SIP providers
    ;                      (instead of type=friend) if you have calls in both directions
      
    
    ;externip =hartwig.ath.cx	; Address that we're going to put in outbound SIP messages
    				; if we're behind a NAT
    
    				; The externip and localnet is used
    				; when registering and communicating with other proxies
    				; that we're registered with
    				; You may add multiple local networks.  A reasonable set of defaults
    				; are:
    localnet=192.238.148.0/255.255.255.0; All RFC 1918 addresses are local networks
    ;localnet=10.0.0.0/255.0.0.0	; Also RFC1918
    ;localnet=172.16.0.0/12		; Another RFC1918 with CIDR notation
    ;localnet=169.254.0.0/255.255.0.0 ;Zero conf local network
    
    ;-----------------------------------------------------------------------------------
    ; Users and peers have different settings available. Friends have all settings,
    ; since a friend is both a peer and a user
    ;
    ; User config options:        Peer configuration:
    ; --------------------        -------------------
    ; context                     context
    ; permit                      permit
    ; deny                        deny
    ; secret                      secret
    ; md5secret                   md5secret
    ; dtmfmode                    dtmfmode
    ; canreinvite                 canreinvite
    ; nat                         nat
    ; callgroup                   callgroup
    ; pickupgroup                 pickupgroup
    ; language                    language
    ; allow                       allow
    ; disallow                    disallow
    ; insecure                    insecure
    ; trustrpid                   trustrpid
    ; progressinband              progressinband
    ; promiscredir                promiscredir
    ; callerid
    ; accountcode
    ; amaflags
    ; incominglimit
    ; restrictcid
    ;                             mailbox
    ;                             username
    ;                             template
    ;                             fromdomain
    ;                             regexten
    ;                             fromuser
    ;                              host
    ;                             mask
    ;                             port
    ;                             qualify
    ;                             defaultip
    ;                             rtptimeout
    ;                             rtpholdtimeout
    
    ;[sip_proxy]
    ;For incoming calls only. Example: FWD (Free World Dialup)
    ;type=friend
    ;context=sipgate.de
    
    [sip_proxy-out]
    type=friend         		; we only want to call out, not be called
    secret=
    username=5559972  Authentication user for outbound proxies
    fromuser=h.mattes.privat		; Many SIP providers require this!
    host=dynamic
    
    ;[grandstream1]
    ;type=friend 			; either "friend" (peer+user), "peer" or "user"
    ;context=from-sip
    ;fromuser=grandstream1		; overrides the callerid, e.g. required by FWD
    ;callerid=John Doe <1234>
    ;host=192.168.0.23		; we have a static but private IP address
    ;nat=no				; there is not NAT between phone and Asterisk
    canreinvite=no		; allow RTP voice traffic to bypass Asterisk
    dtmfmode=rfc2833			; either RFC2833 or INFO for the BudgeTone
    ;incominglimit=1		; permit only 1 outgoing call at a time
    				; from the phone to asterisk
    ;mailbox=1234@default  ; mailbox 1234 in voicemail context "default"
    ;disallow=all			; need to disallow=all before we can use allow=
    ;allow=ulaw			; Note: In user sections the order of codecs
    				; listed with allow= does NOT matter!
    ;allow=alaw
    ;allow=g723.1			; Asterisk only supports g723.1 pass-thru!
    ;allow=g729			; Pass-thru only unless g729 license obtained
    
    
    ;[xlite1]
    ;Turn off silence suppression in X-Lite ("Transmit Silence"=YES)!
    ;Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed
    ;type=friend
    ;regexten=1234                 ; When they register, create extension 1234
    ;username=xlite1
    ;callerid="Jane Smith" <5678>
    ;host=
    ;nat=yes                       ; X-Lite is behind a NAT router
    ;canreinvite=no                ; Typically set to NO if behind NAT
    ;disallow=all
    ;allow=gsm                     ; GSM consumes far less bandwidth than ulaw
    ;allow=ulaw
    ;allow=alaw
    
    
    ;[snom1]
    ;type=friend			; Friends place calls and receive calls
    ;context=sip		; Context for incoming calls from this user
    ;secret=blah
    ;language=de			; Use German prompts for this user 
    ;host=dynamic			; This peer register with us
    ;dtmfmode=inband		; Choices are inband, rfc2833, or info
    ;defaultip=192.238.148.203		; IP used until peer registers
    ;username=snom1			; Username to use in INVITE until peer registers
    ;mailbox=1234,2345		; Mailboxes for message waiting indicator
    ;restrictcid=yes		; To have the callerid restriced -> sent as ANI
    ;disallow=all
    ;allow=ulaw                     ; dtmfmode=inband only works with ulaw or alaw!
    ;mailbox=1234@context,2345      ; Mailbox(-es) for message waiting indicator
    
    
    ;[polycom]
    ;type=friend			; Friends place calls and receive calls
    ;context=from-sip		; Context for incoming calls from this user
    ;secret=blahpoly
    ;host=dynamic			; This peer register with us
    ;dtmfmode=rfc2833		; Choices are inband, rfc2833, or info
    ;username=polly			; Username to use in INVITE until peer registers
    ;disallow=all
    ;allow=ulaw                     ; dtmfmode=inband only works with ulaw or alaw!
    ;progressinband=no		; Polycom phones don't work properly with "never"
    
    
    ;[pingtel]
    ;type=friend
    ;username=pingtel
    ;secret=blah
    ;host=dynamic
    ;insecure=yes			; To match a peer based by IP address only and not peer
    ;insecure=very			; To allow registered hosts to call without re-authenticating
    ;qualify=1000			; Consider it down if it's 1 second to reply
    				; Helps with NAT session
    				; qualify=yes uses default value
    ;callgroup=1,3-4		; We are in caller groups 1,3,4
    ;pickupgroup=1,3-5		; We can do call pick-p for call group 1,3,4,5
    ;defaultip=192.168.0.60		; IP address to use if peer has not registred
    
    ;[cisco1]
    ;type=friend
    ;username=cisco1
    ;secret=blah
    ;qualify=200			; Qualify peer is no more than 200ms away
    ;nat=yes			; This phone may be natted
    				; Send SIP and RTP to  IP address that packet is 
    				; received from instead of trusting SIP headers 
    ;host=dynamic			; This device registers with us
    ;canreinvite=no			; Asterisk by default tries to redirect the
    				; RTP media stream (audio) to go directly from
    				; the caller to the callee.  Some devices do not
    				; support this (especially if one of them is 
    				; behind a NAT).
    ;defaultip=192.168.0.4
    
    ;[cisco2]
    ;type=friend
    ;username=cisco2
    ;fromuser=markster		; Specify user to put in "from" instead of callerid
    ;fromdomain=yourdomain.com	; Specify domain to put in "from" instead of callerid
    				; fromuser and fromdomain are used when Asterisk
    				; places calls to this account.  It is not used for
    				; calls from this account.
    ;secret=blah
    ;host=dynamic
    ;defaultip=192.168.0.4
    ;amaflags=default		; Choices are default, omit, billing, documentation
    ;accountcode=markster		; Users may be associated with an accountcode to ease billing
    
    
    [phone1]
    type=friend
    host=dynamic
    defaultip=192.238.148.215
    port=5060
    username=phone1
    ;secret=blah
    dtmfmode=rfc2833
    mailbox=1000
    context=sip
    callerid="Me"<200>
    
    [phone2]
    type=friend
    host=dynamic
    username=phone2
    ;secret=blah
    defaultip=192.238.148.216
    port=5061
    dtmfmode=rfc2833
    mailbox=1000
    context=sip
    callerid="Mini Me" <210>
    
    [sipgate1]
    type=friend
    username=h.mattes.privat
    secret=
    host=192.238.148.220
    fromuser=h.mattes.privat
    fromdomain=sipgate.de
    context=from sip
    canreinvite=no
    qualify=yes
    disallow=all
    insecure=very
    nat=no
    dtmfmode=rfc2833
    tos=0x18
    callerid="Mattes.privat" <5550815>
    
    [out1]
    type=friend
    host=192.238.148.220
    username=out1
    secret=out1
    context=appout1
    insecure=very
    caninvet=no
    canreinvite=no
    allow=ulaw
    nat=no
    port=5060
    callerid="out1" <123>
    
    und die extension.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 => sipgate
    
    ;
    ; 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. 
    ;
    [sip]
    
    exten => 200,1,Dial(SIP/phone1, 20,tr)
    exten => 210,1,Dial(SIP/phone2, 20,tr)
    exten => 1000,1,Dial(SIP/phone1&SIP/phone2,20,tr)
    
    ;exten => extension number, command priority, command
    
    [sipgate]
    ;Anrufe für sipgate -Account 1 an App. out1 leiten
    exten => sipgate1,1,SetCallerID(${CALLERIDNUM})
    exten => sipgate1,2,Dial(SIP/out1,30,r)
    exten => sipgate1,3,Hangup
    
    [default]
    
    [out1]
    
    exten => _8.,1,Dial(SIP/${EXTEN:1}@sipgate,60,tT)
    exten => _8.,2,Dial,Congestion
    exten => _8.,3,Busy
    exten => _8.,4,Hangup
    
    ichmuss dazu sagen ich hab wirklich keine ahnung wa ich da überhaupt mach, mein chef hat mir das projekt gegeben.. ich bin nur ne praktikantin... hab es mit müh und not bis hier her geschafft..
     
  4. rajo

    rajo Admin-Team

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    Hi,

    ( demnächst code-tags für solch lange configs posten und Kommentare gleich rauslassen, das macht es übersichtlicher ;) )
    Code:
    [sipgate1] 
    ...
    disallow=all 
    
    Jepp, da wäre das erste Problem: Hinter das disallow=all mal ein paar allow= Zeilen machen, je nachdem wie dick die Internetanbindung ist
    z.B. allow=alaw allow=ulaw allow=gsm.

    Und den context=from sip gibt es in Deiner extensions.conf nicht. D.h. da müsste dann context=sip oder context=sipgate stehen -- je nachdem was Du jetzt genau vorhast.

    Hmm... das ist aber keine so gute Voraussetzung :)
    Ggf. hilft das hier weiter: http://www.ippf.tk/forum/viewtopic.php?t=14221
     
  5. Bintangku

    Bintangku Neuer User

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    danke für die Tips , ich hab jetzt alles umgeändert. Jedoch kommen jetzt natürlich weiter Fehlermeldungen....
    und zwar:

    --Apr 19 08:07:43 WARNING[18776]: chan_skinny.c:2584 reload_config: Unable to get our IP address, Skinny disabled
    --Apr 19 08:07:43 WARNING[18776]: chan_oss.c:239 sound_thread: Read error on sound device: Resource temporarily unavailable
    --Apr 19 08:07:50 WARNING[18776]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call 7420d3944337f0d11c346d4c5d04ec4d@192.238.148.220 for seqno 102(Non-critical Request)
    --Apr 19 08:07:50 WARNING[18776]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call 04aca7c44a1a36ab1207478b400673fa@192.238.148.220 for seqno 102(Non-critical Request)
    --Apr 19 08:08:01 WARNING[18776]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call 7eb242695eaf81f9571e67f3064a89ca@192.238.148.220 for seqno 102(Non-critical Request)
     
  6. SeAcabo

    SeAcabo Neuer User

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    also als erstes solltest du mal den titel dieses themas eindeutiger gestalten, denn antworten auch mehr leute :wink:

    zu deinen problemen:
    Skinny ist nicht aktiviert (telefonate nach außen nach innen und umgekehrt sollten funktionieren)

    deine soundkarte auf dem server ist nicht konfiguriert oder es gibt keine oder sie hat gerade was besseres zu tun, sollte allerdings auch kein problem darstellen

    dazu habe ich auch keine lösung parat und hatte es auch eine weile aber es gab keine probleme damit deshalb habe ich es einfach ignoriert