[Erledigt]Brauch Hilfe bei Konfiguration von SIP -> Asterisk -> misdn -> Isdn Telefon

[MP_LEO]

Ich glaube ich brauch nochmal euere Hilfe. Es geht um folgendes:

Erstmal funktioniert momentan die Verbindung zwischen ISDN Telefon und Asterisk.

Damit mir die Konfiguration leichter geht habe ich mir Asterisk-GUI installiert und der funktioniert auch schon soweit. Dort habe ich unter Trunks meine beiden SIP Account eingetragen, die sich auch zu meinem Provider verbinden. Wenn ich jetzt eine der beiden SIP Accounts mit meinem Handy anrufe kommen die auch bis zu meinem Asterisk Server. Nur leider wird mir dann gleich gesagt das der Teilnehmer den Anruf ablehnt (Steht zumindest auf dem Handy). Jetzt habe ich noch einen USER 6000 mit einer SIP Telefonnummer angelegt und will eigentlich das dieser User das ISDN Telefon verwendet. Nur leider geht das irgendwie nicht. Auch bei dem ISDN Telefon ist das problem wenn ich keine Regel erstelle für Telefonate die vom ISDN rein kommen das mir das Telefon sag kein Teilnehmer. Wenn ich haber jetzt eine Regel erstelle für das ISDN Telefon das der Anruf bei drücken der 0 die MailBox abfragen soll geht es. Was mach ich falsch wie kann ich vom ISDN Telefon über ein SIP Account telefonieren und wie kann ich es einstellen das von einem SIP Account die Telefonate an ein ISDN Telefon weitergeleitet werden.

Die Datein wurden am anfang mit sampel erstell darum steht da soviel drin. Die kommentar habe ich mal lieber entfernt.

sip.conf

[general]
context = default
allowoverlap = no
bindport = 5060
bindaddr = 192.168.100.100
srvlookup = yes
allowexternaldomains = yes
allowguest = yes
allowsubscribe = yes
allowtransfer = yes
alwaysauthreject = no
autodomain = no
callevents = no
canreinvite = nonat
checkmwi = 10
compactheaders = no
defaultexpiry = 120
domain = lan
dtmfmode = rfc2833
dumphistory = no
externrefresh = 10
fromdomain = sip.qsc.de
g726nonstandard = no
jbenable = no
jbforce = no
jbimpl = 
jblog = no
jbmaxsize = 
jbresyncthreshold = 
language = 
maxcallbitrate = 384
maxexpiry = 3600
minexpiry = 60
mohinterpret = default
mohsuggest = 
nat = yes
notifyringing = yes
pedantic = yes
progressinband = never
promiscredir = no
realm = asterisk
recordhistory = no
register = user[:secret[:authuser]]@host[:port][/contact]
registerattempts = 0
registertimeout = 20
relaxdtmf = no
rtpholdtimeout = 300
rtptimeout = 60
sendrpid = no
sipdebug = yes
subscribecontext = 
t1min = 100
t38pt_udptl = yes
tos_audio = none
tos_sip = none
tos_video = none
trustrpid = no
useragent = Asterisk PBX
usereqphone = no
videosupport = no
disallow = all
allow = undefined,ulaw,alaw,gsm
[authentication]

users.conf

[general]

fullname = New User

userbase = 6000

hasvoicemail = yes

vmsecret = 1234

hassip = yes

hasiax = yes

hasmanager = no

callwaiting = yes
threewaycalling = yes
callwaitingcallerid = yes
transfer = yes
canpark = yes
cancallforward = yes
callreturn = yes
callgroup = 1
pickupgroup = 1
vmexten = 1111

[12349999990]
context = DID_12349999990
host = sip.qsc.de
trunkname = Firmentelefon  ; GUI metadata
username = 12349999990
secret = password
hasiax = no
registeriax = no
hassip = yes
registersip = yes
trunkstyle = voip
hasexten = no
fromdomain = sip.qsc.de
fromuser = 12349999990
insecure = port,invite
authuser = 12349999990
disallow = all
allow = ulaw,alaw,gsm,g726

[6000]
username = 6000
transfer = yes
mailbox = 6000
call-limit = 100
fullname = 12349999990
registersip = no
host = dynamic
callgroup = 1
context = DLPN_DialPlan1
cid_number = 6000
hasvoicemail = yes
vmsecret = 1234
email = 
threewaycalling = no
hasdirectory = no
callwaiting = no
hasmanager = no
hasagent = no
hassip = yes
hasiax = no
secret = 
nat = yes
canreinvite = yes
dtmfmode = rfc2833
insecure = no
pickupgroup = 1
autoprov = yes
label = 6000
macaddress = :
linenumber = 1
LINEKEYS = 1
disallow = all
allow = ulaw,gsm

[trunk_1]
context = DID_trunk_1
host = sip.qsc.de
username = 12349999991
secret = password
trunkname = FAX  ; GUI metadata
hasiax = no
registeriax = no
hassip = yes
registersip = yes
trunkstyle = voip
hasexten = no
insecure = no
fromdomain = sip.qsc.de
fromuser = 12349999991
disallow = all
allow = ulaw,alaw,gsm,g726

[6001]
username = 6001
transfer = yes
mailbox = 6001
call-limit = 100
fullname = 12349999991
registersip = no
host = dynamic
callgroup = 1
context = DLPN_DialPlan1
cid_number = 6001
hasvoicemail = no
vmsecret = 
email = 
threewaycalling = no
hasdirectory = no
callwaiting = no
hasmanager = no
hasagent = no
hassip = no
hasiax = yes
secret = 
nat = yes
canreinvite = no
dtmfmode = rfc2833
insecure = no
pickupgroup = 1
disallow = all
allow = ulaw,gsm
autoprov = no
label = 
macaddress = 
linenumber = 1
LINEKEYS = 1

[trunk_2]
context = DID_trunk_2
host = 127.0.0.1
username = iaxmodem
secret = password
trunkname = IAX-Modem  ; GUI metadata
hasiax = yes
registeriax = yes
hassip = no
registersip = no
trunkstyle = voip
hasexten = no
disallow = all
allow = all

misdn.conf

[general]
misdn_init = /etc/misdn-init.conf
debug = 0
ntdebugflags = 0
ntdebugfile = /var/log/misdn-nt.log
ntkeepcalls = no
bridging = no
l1watcher_timeout = 0
stop_tone_after_first_digit = yes
append_digits2exten = yes
dynamic_crypt = no
crypt_prefix = **
crypt_keys = test,muh
[default]
context = misdn
language = en
musicclass = default
senddtmf = yes
far_alerting = no
allowed_bearers = all
nationalprefix = 0
internationalprefix = 00
rxgain = 0
txgain = 0
te_choose_channel = no
pmp_l1_check = no
reject_cause = 16
need_more_infos = no
nttimeout = no
method = standard
overlapdial = yes
dialplan = 0
localdialplan = 0
cpndialplan = 0
early_bconnect = yes
incoming_early_audio = no
nodialtone = no
presentation = -1
screen = -1
echotraining = no
jitterbuffer = 4000
jitterbuffer_upper_threshold = 0
hdlc = no
max_incoming = -1
max_outgoing = -1
[intern]
ports = 1,2
context = Intern
[internPP]
ports = 3
[first_extern]
ports = 4
context = Extern1
msns = *
[second_extern]
ports = 5
context = Extern2
callerid = 15
msns = 102,144,101,104
[trunk_m1]
trunkname = FirmentelefonISDN
context = DID_trunk_m1
ports = 1
hasmisdn = yes
msns = *

extensions.conf

[general]
static = yes
writeprotect = no
clearglobalvars = no
[globals]
CONSOLE = Console/dsp  ; Console interface for demo
IAXINFO = guest  ; IAXtel username/password
TRUNK = Zap/G2  ; Trunk interface
TRUNKMSD = 1  ; MSD digits to strip (usually 1 or 0)
FEATURES = 
DIALOPTIONS = tThHkK
RINGTIME = 20
FOLLOWMEOPTIONS = 
PAGING_HEADER = Intercom
PAGING_TIMEOUT = 60
12349999990 = SIP/12349999990
CID_12349999990 = 12349999990
CID_6000 = 12349999990
trunk_m1 = mISDN/g:trunk_m1
trunk_1 = SIP/trunk_1
CID_trunk_1 = 12349999991
CID_6001 = 12349999991
trunk_2 = IAX2/trunk_2
[dundi-e164-canonical]
[dundi-e164-customers]
[dundi-e164-via-pstn]
[dundi-e164-local]
include => dundi-e164-canonical
include => dundi-e164-customers
include => dundi-e164-via-pstn

[dundi-e164-switch]
switch => DUNDi/e164

[dundi-e164-lookup]
include => dundi-e164-local
include => dundi-e164-switch

[macro-dundi-e164]
exten => s,1,Goto(${ARG1},1)
include => dundi-e164-lookup

[iaxtel700]
exten => _91700XXXXXXX,1,Dial(IAX2/${GLOBAL(IAXINFO)}@iaxtel.com/${EXTEN:1}@iaxtel)

[iaxprovider]
[trunkint]
exten => _9011.,1,Macro(dundi-e164,${EXTEN:4})
exten => _9011.,n,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})

[trunkld]
exten => _91NXXNXXXXXX,1,Macro(dundi-e164,${EXTEN:1})
exten => _91NXXNXXXXXX,n,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})

[trunklocal]
exten => _9NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})

[trunktollfree]
exten => _91800NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
exten => _91888NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
exten => _91877NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
exten => _91866NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})

[international]
ignorepat => 9
include => longdistance
include => trunkint

[longdistance]
ignorepat => 9
include => local
include => trunkld

[local]
ignorepat => 9
include => default
include => trunklocal
include => iaxtel700
include => trunktollfree
include => iaxprovider
include => parkedcalls

[macro-trunkdial]
exten => s,1,Dial(${ARG1})
exten => s,n,Goto(s-${DIALSTATUS},1)
exten => s-NOANSWER,1,Hangup
exten => s-BUSY,1,Hangup
exten => _s-.,1,NoOp

[macro-stdPrivacyexten];
exten => s,1,Dial(${ARG2},20|p)  ; Ring the interface, 20 seconds maximum, call screening 
exten => s,2,Goto(s-${DIALSTATUS},1)  ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten => s-NOANSWER,1,Voicemail(${ARG1},u)  ; If unavailable, send to voicemail w/ unavail announce
exten => s-NOANSWER,2,Goto(default,s,1)  ; If they press #, return to start
exten => s-BUSY,1,Voicemail(${ARG1},b)  ; If busy, send to voicemail w/ busy announce
exten => s-BUSY,2,Goto(default,s,1)  ; If they press #, return to start
exten => s-DONTCALL,1,Goto(${ARG3},s,1)  ; Callee chose to send this call to a polite "Don't call again" script.
exten => s-TORTURE,1,Goto(${ARG4},s,1)  ; Callee chose to send this call to a telemarketer torture script.
exten => _s-.,1,Goto(s-NOANSWER,1)  ; Treat anything else as no answer
exten => a,1,VoicemailMain(${ARG1})  ; If they press *, send the user into VoicemailMain

[macro-page];
exten => s,1,ChanIsAvail(${ARG1}|js)  ; j is for Jump and s is for ANY call
exten => s,n,GoToIf([${AVAILSTATUS} = "1"]?autoanswer:fail)
exten => s,n(autoanswer),Set(_ALERT_INFO="RA")  ; This is for the PolyComs
exten => s,n,SIPAddHeader(Call-Info: Answer-After=0)  ; This is for the Grandstream, Snoms, and Others
exten => s,n,NoOp()  ; Add others here and Post on the Wiki!!!!
exten => s,n,Dial(${ARG1}||)
exten => s,n(fail),Hangup

[demo]
exten => s,1,Wait(1)  ; Wait a second, just for fun
exten => s,n,Answer  ; Answer the line
exten => s,n,Set(TIMEOUT(digit)=5)  ; Set Digit Timeout to 5 seconds
exten => s,n,Set(TIMEOUT(response)=10)  ; Set Response Timeout to 10 seconds
exten => s,n(restart),BackGround(demo-congrats)  ; Play a congratulatory message
exten => s,n(instruct),BackGround(demo-instruct)  ; Play some instructions
exten => s,n,WaitExten  ; Wait for an extension to be dialed.
exten => 2,1,BackGround(demo-moreinfo)  ; Give some more information.
exten => 2,n,Goto(s,instruct)
exten => 3,1,Set(LANGUAGE()=fr)  ; Set language to french
exten => 3,n,Goto(s,restart)  ; Start with the congratulations
exten => 1000,1,Goto(default,s,1)
exten => 1234,1,Playback(transfer,skip)  ; "Please hold while..." 
exten => 1234,n,Macro(stdexten,1234,${GLOBAL(CONSOLE)})
exten => 1235,1,Voicemail(1234,u)  ; Right to voicemail
exten => 1236,1,Dial(Console/dsp)  ; Ring forever
exten => 1236,n,Voicemail(1234,b)  ; Unless busy
exten => #,1,Playback(demo-thanks)  ; "Thanks for trying the demo"
exten => #,n,Hangup  ; Hang them up.
exten => t,1,Goto(#,1)  ; If they take too long, give up
exten => i,1,Playback(invalid)  ; "That's not valid, try again"
exten => 500,1,Playback(demo-abouttotry)  ; Let them know what's going on
exten => 500,n,Dial(IAX2/[email protected]/s@default)  ; Call the Asterisk demo
exten => 500,n,Playback(demo-nogo)  ; Couldn't connect to the demo site
exten => 500,n,Goto(s,6)  ; Return to the start over message.
exten => 600,1,Playback(demo-echotest)  ; Let them know what's going on
exten => 600,n,Echo  ; Do the echo test
exten => 600,n,Playback(demo-echodone)  ; Let them know it's over
exten => 600,n,Goto(s,6)  ; Start over
exten => 76245,1,Macro(page,SIP/Grandstream1)
exten => _7XXX,1,Macro(page,SIP/${EXTEN})
exten => 7999,1,Set(TIMEOUT(absolute)=60)
exten => 7999,2,Page(Local/Grandstream1@page&Local/Xlite1@page&Local/1234@page/n|d)
exten => 8500,1,VoicemailMain
exten => 8500,n,Goto(s,6)

[page]
exten => _X.,1,Macro(page,SIP/${EXTEN})

[default]
exten = _#6XXX,1,Set(MBOX=${EXTEN:1}@default)
exten = _#6XXX,n,VoiceMail(${MBOX})
exten = a,1,VoicemailMain(${MBOX})
exten = 1111,1,VoiceMailMain(${CALLERID(num)}@default)

[macro-stdexten-followme]
exten = s,1,Answer
exten = s,2,Dial(${ARG2},${RINGTIME},${DIALOPTIONS})
exten = s,3,Set(__FMCIDNUM=${CALLERID(num)})
exten = s,4,Set(__FMCIDNAME=${CALLERID(name)})
exten = s,5,Followme(${ARG1},${FOLLOWMEOPTIONS})
exten = s,6,Voicemail(${ARG1},u)
exten = s-NOANSWER,1,Voicemail(${ARG1},u)
exten = s-BUSY,1,Voicemail(${ARG1},b)
exten = s-BUSY,2,Goto(default,s,1)
exten = _s-.,1,Goto(s-NOANSWER,1)
exten = a,1,VoicemailMain(${ARG1})

[conferences]

[ringgroups]
exten = 6400,1,Goto(ringroups-custom-1,s,1)

[queues]
exten = 6500,1,Queue(${EXTEN})

[voicemenus]

[voicemailgroups]

[directory]

[page_an_extension]

[pagegroups]

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

[macro-trunkdial-failover-0.3]
exten = s,1,GotoIf($[${LEN(${FMCIDNUM})} > 6]?1-fmsetcid,1)
exten = s,2,GotoIf($[${LEN(${GLOBAL_OUTBOUNDCIDNAME})} > 1]?1-setgbobname,1)
exten = s,3,Set(CALLERID(num)=${IF($[${LEN(${CID_${CALLERID(num)}})} > 2]?${CID_${CALLERID(num)}}:)})
exten = s,n,GotoIf($[${LEN(${CALLERID(num)})} > 6]?1-dial,1)
exten = s,n,Set(CALLERID(all)=${IF($[${LEN(${CID_${ARG3}})} > 6]?${CID_${ARG3}}:${GLOBAL_OUTBOUNDCID})})
exten = s,n,Goto(1-dial,1)
exten = 1-setgbobname,1,Set(CALLERID(name)=${GLOBAL_OUTBOUNDCIDNAME})
exten = 1-setgbobname,n,Goto(s,3)
exten = 1-fmsetcid,1,Set(CALLERID(num)=${FMCIDNUM})
exten = 1-fmsetcid,n,Set(CALLERID(name)=${FMCIDNAME})
exten = 1-fmsetcid,n,Goto(1-dial,1)
exten = 1-dial,1,Dial(${ARG1})
exten = 1-dial,n,Gotoif(${LEN(${ARG2})} > 0 ?1-${DIALSTATUS},1:1-out,1)
exten = 1-CHANUNAVAIL,1,Dial(${ARG2})
exten = 1-CHANUNAVAIL,n,Hangup()
exten = 1-CONGESTION,1,Dial(${ARG2})
exten = 1-CONGESTION,n,Hangup()
exten = 1-out,1,Hangup()

[DID_12349999990]
include = DID_12349999990_default

[DID_12349999990_default]
exten = _X,1,Goto(ringroups-custom-1,s,1)

[macro-stdexten]
exten = s,1,Set(__DYNAMIC_FEATURES=${FEATURES})
exten = s,2,GotoIf($["${FOLLOWME_${ARG1}}" = "1"]?5:3)
exten = s,3,Dial(${ARG2},${RINGTIME},${DIALOPTIONS})
exten = s,4,Goto(s-${DIALSTATUS},1)
exten = s,5,Macro(stdexten-followme,${ARG1},${ARG2})
exten = s-NOANSWER,1,Voicemail(${ARG1},u)
exten = s-NOANSWER,2,Goto(default,s,1)
exten = s-BUSY,1,Voicemail(${ARG1},b)
exten = s-BUSY,2,Goto(default,s,1)
exten = _s-.,1,Goto(s-NOANSWER,1)
exten = a,1,VoicemailMain(${ARG1})

[macro-pagingintercom]
exten = s,1,SIPAddHeader(Alert-Info: ${PAGING_HEADER})
exten = s,2,Page(${ARG1}|${ARG2})
exten = s,3,Hangup

[CallingRule_Firmentelefon]
exten = _i,1,Macro(trunkdial-failover-0.3,${trunk_m1}/${EXTEN:0},,trunk_m1,trunk_m1)

[DLPN_DialPlan1]
include = CallingRule_Firmentelefon
include = default
include = parkedcalls
include = conferences
include = ringgroups
include = voicemenus
include = queues
include = voicemailgroups
include = directory
include = pagegroups
include = page_an_extension

[DID_trunk_m1]
include = DID_trunk_m1_default

[DID_trunk_1]
include = DID_trunk_1_default

[DID_trunk_1_default]
exten = 12349999990,1,Voicemail(6000,u)

[DID_trunk_2]
include = DID_trunk_2_default

[DID_trunk_2_default]
exten = _0,1,Goto(default,6001,1)

[DID_trunk_m1_default]

[CallingRule_FAX]
exten = _X,1,Macro(trunkdial-failover-0.3,${trunk_2}/${EXTEN:0},,trunk_2,)

[ringroups-custom-1]
exten = s,1,NoOp(Firmentelefon)
exten = s,n,Dial(SIP/6000,30,${DIALOPTIONS}i)
exten = s,n,Voicemail(6000,u)

 

[MP_LEO]

hat siche erstmal erledigt bin auf FreePBX umgestiegen