[Asterisk-Users] Anybody managed to call a phone through sipgate.de

Birk Bremer birk.bremer at web.de
Fri Feb 27 09:46:53 MST 2004


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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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