[Asterisk-Users] Anybody managed to call a phone through sipgate.de
David J Carter
david.carter at codepipe.com
Fri Feb 27 10:14:38 MST 2004
Hi,
I would be tempted to get rid of the slash and number on the register line,
unless your asterisk extension is 02115800XXXX.
dave
-----Original Message-----
From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com]On Behalf Of Birk Bremer
Sent: 27 February 2004 16:47
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] Anybody managed to call a phone through
sipgate.de
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Hello everybody,
has anybody managed to call a (old fashioned) phone using Sipgate.de and
asterisk? (yes I have money on my account :-) )
The configuration I got from the sipgate.de people is at the botton of
the mail
Here is mine:
sip.conf:
register => 800XXXX:SECRET at sipgate.de/02115800XXXX
[sipgate]
type=friend
username=800XXXX
secret=SECRET
host=sipgate.de
fromuser=800XXXX
fromdomain=sipgate.net
nat=no
;dtmfband=3Dinband
context=sipin
canreinvite=no
extension.conf:
exten => _9.,1,Dial(SIP/${EXTEN:1}@sipgate.de,30,tr)
To be called on my sipgate number - no problem
If I want to call somebody I get the following error:
When I call a number directly out of the softphone:
Executing Dial("IAX2[myself at myself]/2", "SIP/number at sipgate.de|30|tr")
in new stack
~ -- Called number at sipgate.de
~ -- Got SIP response 403 "Forbidden" back from 217.10.79.9
~ == No one is available to answer at this time
~ -- Hungup 'IAX2[myself at myself]/2
when I use the webinterface at sipgate.de I get a ring at my softphone,
when I pick the call I get the message (in the appearing box)
"Teilnehmer nicht gefunden" - User/Number not found
sometimes (while tried different config. I also got (at * console) to
many hops...
Has anybody managed this - can you please send me your configuration
(sip, extensions) .... or can anybody help
Thanks in advance
Birk Bremer
The configuration the sipgate people suggest:
~ > register => 800XXXX:sipgatepasswort at sipgate.de/800XXXX
^^^^^ can't be correct
|
|
|
| [sipgate]
|
| type=friend
|
| username=800XXXX
|
| secret=sipgatepasswort
|
| host=sipgate.de
|
| fromuser=800XXXX
|
| fromdomain=sipgate.net
|
| nat=yes
|
| ;dtmfband=inband
|
| context=incomingsipgate
|
| canreinvite=no
|
|
|
| Aus der extensions.conf :
|
|
|
| [incomingsipgate]
|
| exten => h,1,Hangup
|
| exten => 800XXXX,1,Dial(SIP/internestelefon,20,tr)
|
|
|
| [sipgate]
|
| exten => _9.,1,Dial(SIP/${EXTEN:1}@sipgate,30,tr)
|
| exten => _9.,2,Playback(invalid)
|
| exten => _9.,3,Hangup
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