Sipgate Account auf einen lokalen SIP User mappen

[kperas]

Hallo Forum,
ich hab da ein kleines Problem und finde einfach kein Beispiel.

Ich habe ein paar lokale SIP User, so im Bereich 10-20. Diese können untereinander telefonieren und über CAPI raustelefonieren.
Ich habe eine Fritzcard, welche an der MSN 109 unserer Telefonanlage hängt, ja halt CAPI eben...
Jetzt habe ich mein * noch bei sipgate registriert und möchte, wenn calls auf diesen Account kommen sie einem bestimmten SIP User [14] zuordnen.
Wie geht sowas?
Hier noch meine Daten:

sipconf:

[general]
context=default ; Default context for incoming calls
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
disallow=all ; First disallow all codecs
;allow=g729
allow=gsm
allow=alaw
allow=ulaw ; Allow codecs in order of preference
; This may also be set for individual users/peers
language=de ; Default language setting for all users/peers

register=>SIPID[email protected]/SIPID
nat=no
canreinvite=no
tos=0x18
insecure=very
nat=yes
dtmfmode=info
maxexpirey=3600
defaultexpirey=600
port=5060
bindaddr=0.0.0.0
localnet=172.22.0.0/255.255.0.0

[sipgate]
type=friend
username=SIPID
secret=XXXXXXXX
host=sipgate.de
fromuser=XXXXXXX
fromdomain=sipgate.de
canreinvite=no
qualify=no
disallow=all
allow=ulaw
allow=alaw
allow=ilbc
allow=gsm
;allow=g729
insecure=very
nat=yes
dtmfmode=info
tos=0x18

[10] ; kphone 2
type=friend
username=10
dtmfmode=rfc2833
secret=10
host=dynamic
callerid="10"= <10>
disallow=all
allow=alaw
allow=ulaw

;[11] ; bettschnitt SIP Telefon
;type=friend
;username=11
;secret=11
;host=172.22.20.152
;callerid="11"= <11>

[12] ;windows Rechner Peras mit SIPP
type=friend
username=12
secret=12
host=dynamic
callerid="12"= <12>

[14] ; SIP Telefon Peras
type=friend
username=14
secret=14
host=dynamic
callerid="14"= <14>

[15] ; SIP Telefon Bettschnitt
type=friend
username=15
secret=15
host=dynamic
callerid="15"= <15>

;[16] ; SIP Telefon Dummy
;type=friend
;username=16
;secret=16
;host=dynamic
;callerid="16"= <16>

;[12]
;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=12 ; When they register, create extension 1234
;username=12
;callerid="12"= <12>
;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

[17] ; kphone 1
type=friend
username=17
secret=17
host=dynamic
dtmfmode=inband
callerid="17"= <17>

[18] ; kphone Demleidb
type=friend
username=18
secret=18
host=dynamic
dtmfmode=inband
callerid="18"= <18>

[19] ; minisip
type=friend
username=19
secret=19
host=dynamic
dtmfmode=inband
callerid="19"= <19>

[20]
type=friend
username=20
secret=20
host=dynamic
callerid="20"= <20>
disallow=all
allow=ulaw
allow=alaw


extensions.conf:

[general]
static=yes
writeprotect=no
[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/userass@provider

;[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 www.gnophone.com or www.iaxtel.com
;
[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/userassword@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/[email protected]/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
include => sipgatecalls

;
; 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/${EXTEN:2}@sipprovider,,r)
;exten => _42X.,1,Dial(SIP/userasswd@${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/[email protected])
;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.
;
include=> 10
;include=> 11
include=> 12
include=> 14
include=> 15
;include=> 16
include=> 17
include=> 18
include=> 19
include=> 20
include=> 99

exten=>_xx.,1,Dial,CAPI/109:${EXTEN}
exten=>_xx.,2,Congestion

[capicall]
exten=>109,1,Dial(SIP/14)
exten=>110,1,Dial(SIP/20)


[10]
;exten => 10,1,Dial(SIP/${EXTEN},60)
exten => 10,1,Dial(SIP/10)
exten => 10,2,Hangup
exten => 10,102,Busy

;[11]
;exten => 11,1,Dial(SIP/11)
;exten => 11,2,Hangup
;exten => 11,102,Busy

[12]
;exten => 12,1,Dial(SIP/12)
exten => 12,1,Dial(SIP/12,10)
exten => 12,2,Dial(SIP/15,10)
exten => 12,3,Answer
exten => 12,4,Playback(vm-options)
exten => 12,5,Hangup
exten => 12,102,Busy

[14]
exten => 14,1,Dial(SIP/14)
exten => 14,2,Hangup
exten => 14,102,Busy

[15]
exten => 15,1,Dial(SIP/15)
exten => 15,2,Hangup
exten => 15,102,Busy

[17]
exten => 17,1,Dial(SIP/17)
exten => 17,2,Hangup
exten => 17,102,Busy

[18]
exten => 18,1,Dial(SIP/18)
exten => 18,2,Hangup
exten => 18,102,Busy

[19]
exten => 19,1,Dial(SIP/19)
exten => 19,2,Hangup
exten => 19,102,Busy

[20]
exten => 20,1,Dial(SIP/20,20)
exten => 20,2,Hangup
exten => 20,102,Busy

[99]
exten => 99,1,Dial,CAPI/109:09001001191 ;Zeitansage der DTelekom

[sipgate]
exten => 555xxxx,1,Dial(SIP/14,20)


MfG
Klaus

 

[WrMulf]

das nächste mal Kommentarzeilen bitte entfernen, das ist übersichtlicher.

Der Ansatz ist nicht schlecht, du hast im default-context der extension.conf ein:

include => sipgatecalls

allerdings gibt es diesen context nirgends. Du brauchst noch:

[sipgatecalls]
exten => SIPID,1,Dial(SIP/14)
exten => SIPID,2,Hangup

und in dem context "[sipgate]" in der sip.conf fehlt noch eine Zeile:

context=sipgatecalls

Damit sollte es gehen.