sip.conf for sipgate.de:
-----
[general]
dtmfmode=inband
port = 5060
bindaddr = 192.168.254.204
context = incoming
tos=0x18
maxexpirey=1800
defaultexpirey=300
disallow=all
allow=ulaw
nat=yes
register => 8001234:[EMAIL PROTECTED]/99049
[sip99049]
port=5060
secret=secret
username=8001234
;authuser=8001234
fromuser=8001234
fromdomain=sipgate.net
type=friend
host=sipgate.de
nat=yes
dtmfmode=rfc2833
canreinvite=no
context=incoming
callerid="Aber Hallo" <0211 58001234>
-----
And from extensions.conf:
-----
; 01149 (Germany) dialed through sipgate.de
[dial011]
exten => _01149.,1,Dial(SIP/0${EXTEN:[EMAIL PROTECTED],120)
exten => _01149.,2,Congestion
[incoming]
exten => 99049,1,SetCIDName(From Germany)
exten => 99049,2,Macro(dialext_incoming,${EXT_ALL},1000)
-----
Works like a charm, all the time. I have ten extensions on five
SPA-2000 and one IP-Phone.
To the OP: Don't bother with the ISDN phone -- the cost of the ISDN card
just isn't worth it. Get yourself a Sipura, and you can even do
distinctive ring and such.
-----Original Message-----
From: box100 [mailto:[EMAIL PROTECTED]
Sent: Wednesday, July 21, 2004 8:53 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Help needed for Seting Up Asterisk
Ich habe eine einfachere Konfiguration IAX �ber Asterisk mit einem
Sipura SPA-2000-Ger�t, dass sich mit der Asterisk registriert, aber ich
habe bisher keinen Erfolg gehabt, dadurch eine Sipgate-Verbindungen
aufzustellen.
Hast Du es mal mit IAX versucht?
Gruss,
Roger (aus Michigan)
From:Beierlein Moritz [mailto:[EMAIL PROTECTED]
Sent:Wed 7/21/2004 15:04
To:[EMAIL PROTECTED]
Subject:[Asterisk-Users] Help needed for Seting Up Asterisk
Hello List,
I'm from Germany and I want to use a Asterisk System.
I have a few Accounts at my SIP-Provider www.sipgate.de and now I want
to use my ISDN-Phone on the Sip-System.
My idea was i set up a Asterisk-System and i will put in an ISDN Card
where I can plug a ISDN Phone, I will have to use an ISDN card with the
NT-Mode.
The Asterisk has to register is at the SIP Provider and if a Call comes
to me the Asterisk has to gibe the call to the ISDN card where the
Telephone will ring.
If the SIP Account 1 rings the telephone should get the MSN 1 and if
Account 2 rings, the telephone should get the MSN 2.
I will use Asterisk behind a NAT Router. If the Internetconnection
interrupts the Asterisk has to wait 20 seconds, then has to register at
the SIP-Provider.
How can I do this, can somebody please help me?
How is it possible to get the SIP Calls to the ISDN card?
Would be very nice if you could help me.
Thanks
Moritz Beierlein
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