Der Rechner ist mit der FritzCard zur Telefonanlage verbunden. hat intern die nummer "1615". intern kann ich faxe versenden, funktioniert. nach aussen kann er auch schicken und ist auch von aussen zu erreichen.
ich möchte halt, das ich von meinem arbeitsplatz-pc aus mit anderen leuten über den server raus telefonieren kann, und eben anders herum.
hier meine SIP.conf
;
; 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 ; 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=184 ; 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=ulaw ; 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=en ; 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
;promiscredir = no ; If yes, allows 302 or REDIR to non-local SIP address
; Asterisk can register as a SIP user agent to a SIP proxy (provider)
; Format for the register statement is:
; register => user[:secret[:authuser]]@host[
ort][/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
[email protected]
;
; This will pass incoming calls to the 's' extension
;
;
;register => 2345
assword@sip_proxy/1234
;
; 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 = 200.201.202.203 ; 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=192.168.0.0/255.255.0.0; 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
; auth auth
; 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
; fromuser
; host
; mask
; port
; qualify
; defaultip
; rtptimeout
; rtpholdtimeout
;[sip_proxy]
; For incoming calls only. Example: FWD (Free World Dialup)
;type=user
;context=from-fwd
;[sip_proxy-out]
;type=peer ; we only want to call out, not be called
;secret=guessit
;username=yourusername ; Authentication user for outbound proxies
;fromuser=yourusername ; Many SIP providers require this!
;host=box.provider.com
;[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=yes ; allow RTP voice traffic to bypass Asterisk
;dtmfmode=info ; 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
;username=xlite1
;callerid="Jane Smith" <5678>
;host=dynamic
;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
;[snom]
;type=friend ; Friends place calls and receive calls
;context=from-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=192.168.0.59 ; IP used until peer registers
;username=snom ; 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
;[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
und meine extensions.conf
; Sample /etc/asterisk/extensions.conf
; Created September 1, 2004
; =========================================================
; QUICKSTART WITH VOICEPULSE CONNECT! SERVICE:
;
; * Login to your VoicePulse Connect! account at:
;
http://connect.voicepulse.com/
;
; * Go to the Devices tab and note your device login and
; password
;
; * Replace MY_DEVICE_LOGIN and MY_DEVICE_PASSWORD in the
; "exten => " statements below with your device login
; and password. (Lines 81-82)
;
; * If you DO NOT have a phone number from VoicePulse
; Connect!, comment out the following lines by placing a
; semicolon ";" at the beginning:
;
; - The entire "[arbitrary-name]" context (lines 43-48)
; - The entire "[testdtmf]" context (lines 54-60)
; =========================================================
[general]
static=yes
writeprotect=no
[globals]
; ---------------------------------------------------------
; [arbitrary-name] is the context referred to by the
; [voicepulse-in-01] user in iax.conf. This is where your
; custom incoming call processing should go.
;
; For sample purposes, this section will read back the
; dialed number and then test DTMF by reading back each
; digit pressed by the caller.
; ---------------------------------------------------------
[arbitrary-name] ; <-- Should match the context you have
; under [voicepulse-in-01] in iax.conf
exten => _NXXNXXXXXX,1,Playback(beep)
exten => _NXXNXXXXXX,2,SayDigits(${EXTEN})
exten => _NXXNXXXXXX,3,Goto(testdtmf|s|1)
;
; This context is used by the sample [arbitrary-name]
; context above to read back each digit you press.
;
[testdtmf]
exten => s,1,Background(beep)
exten => s,2,ResponseTimeout(60)
exten => _x,1,SayDigits(${EXTEN})
exten => _x,2,Goto(testdtmf|s|1)
exten => i,1,Goto(testdtmf|s|1)
exten => t,1,Hangup
; ---------------------------------------------------------
; This context is used to send all outgoing calls to the
; VoicePulse Connect! service for connection to the PSTN.
;
; Asterisk will attempt to dial out through gwiaxt01 first.
; If there is a problem, it will attempt to dial out
; through gwiaxt02.
;
; YOU MUST HAVE BOTH LINES FOR OUTGOING CALL REDUNDANCY!
;
; ---------------------------------------------------------
;
; There should be TWO lines after [outgoing], each beginning
; with "exten =>". Please check to make sure copying or
; cutting & pasting this sample did not break the lines into
; more than TWO exten lines.
;
[outgoing]
exten => _1NXXNXXXXXX,1,Dial(IAX2/MY_DEVICE_LOGIN:
[email protected]/${EXTEN})
exten => _1NXXNXXXXXX,102,Dial(IAX2/MY_DEVICE_LOGIN:
[email protected]/${EXTEN})
Vielleicht kannst Du mir da ja helfen... dass das richtig funktioniert.
MfG
Cosmos
