Pascal Speck
2005-Sep-07 09:28 UTC
[Asterisk-Users] Second Line does not Connect - HELP - misdn,sip
About my System:
2 * HFC Cards with misdn. 1 NT mode, 1 TE mode
1 * Sip-Provider (1und1)
On NT-Port --> Ritto (Elmeg) PBX
On TE-Port --> NTBA
About my Problem:
When a SIP-Call from a phone connected to the Ritto PBX is in progress and
someone calls on the ISDN-Line, the greeting works, and the phones connected
to the Ritto PBX are ringing. When I pick up a phone there is no connection
and the caller hears MOH all the time. This only happens when a second call
is in progress.
When no other call is in progress, everything works fine.
About my Configfiles:
extensions.conf
[incoming]
exten => xxxx,1,Goto(anruferannahme,s,1)
exten => xxxx,1,Goto(anruferannahme,s,1)
exten => 922xxx,1,Answer()
exten => 922xxx,2,Dial(misdn/2/922975) ; FAX
exten => 923xxx,1,Answer()
exten => 923xxx,2,Playback(thomas)
exten => 923xxx,3,Dial(misdn/2/9230250,,m) ; Thomas Durchwahl
exten => 923xxx,1,Answer()
exten => 923xxx,2,Dial(misdn/2/9230251) ; Thomas FAX
[outgoing]
; Anwahl ?ber normale ISDN-Leitung:
exten => _999.,1,Answer()
exten => _999.,2,Dial(misdn/1/${EXTEN:3},,m)
exten => _999.,3,Playback(dialfailed)
; Faxe ?ber normalen ISDN-Anschluss verschicken:
exten => _X./922975,1,WaitforDigits(2000) ; mit Vorwahl
exten => _X./922975,2,Answer()
exten => _X./922975,3,Dial(misdn/1/${EXTEN}) ; wenn IP nich erfolgreich
; Telefongespr?che bei denen die Vorwahl angegeben ist:
exten => _0X.,1,WaitforDigits(4000)
exten => _0X.,2,Answer()
exten => _0X.,3,Dial(SIP/${EXTEN}@sip.1und1.de)
exten => _0X.,4,Playback(nosip)
exten => _0X.,5,Dial(misdn/1/${EXTEN}) ; wenn IP nicht erfolgreich
exten => _0X.,6,Playback(dialfailed)
exten => _0X.,104,Playback(besetzt)
; Telefongespr?che bei denen die Vorwahl nicht angegeben ist:
exten => _X.,1,WaitforDigits(4000)
exten => _X.,2,Answer()
exten => _X.,3,Dial(SIP/02774${EXTEN}@sip.1und1.de)
exten => _X.,4,Playback(nosip)
exten => _X.,5,Dial(misdn/1/${EXTEN})
exten => _X.,6,Playback(dialfailed)
exten => _X.,104,Playback(besetzt)
[aufnahme]
exten => s,1,Background(beep)
exten => 1,1,Record(/var/lib/asterisk/sounds/greeting:gsm)
exten => 2,1,Record(/var/lib/asterisk/sounds/besetzt:gsm)
exten => 3,1,Record(/var/lib/asterisk/sounds/aufnahme:gsm)
[anruferannahme]
exten => s,1,Answer()
exten => s,2,Background(greeting)
exten => s,3,Dial(misdn/2/4444,15,m)
;exten => s,4,WaitMusicOnHold(2)
;exten => s,5,Dial(misdn/2/9230255,15,m)
;exten => s,6,WaitMusicOnHold(2)
;exten => s,7,Dial(misdn/2/4444,100,m)
;exten => s,8,Playback(nichterr)
exten => s,4,Hangup()
exten => 7,1,Goto(aufnahme,s,1)
misdn.conf
[general]
context=vs
language=de
immediate=yes
debug=2
allow=alaw
musiconhold=default
[TEport]
context=incoming
ports=1
msns=*
[NTport]
context=outgoing
ports=2
sip.conf
[general]
port = 5060
bindaddr = 0.0.0.0
externip = myip
localnet = 192.168.0.0/255.255.0.0
context = default
srvlookup = yes
disallow = all
allow = ulaw
nat = yes
register => 492774xxxx:mysecret@sip.1und1.de/492774xxxx
[sip.1und1.de]
type=friend
username=492774xxxx
fromuser=492774xxxx
secret=mysecret
host=sip.1und1.de
context=incoming
fromdomain=1und1.de
qualify=no
insecure=very
canreinvite=no
nat=yes
allow=g726
dtmfmode=info
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