[Asterisk-Users] Second Line does not Connect - HELP - misdn,sip

[Pascal Speck]

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 über normale ISDN-Leitung:

exten => _999.,1,Answer()

exten => _999.,2,Dial(misdn/1/${EXTEN:3},,m)

exten => _999.,3,Playback(dialfailed)

 

; Faxe über 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

 

; Telefongespräche 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)

 

; Telefongespräche 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 at 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

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050907/7f30ad06/attachment.htm