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

[Sascha Knific]

Hi Birk

I´m messing arround for the last 2 day with sipgate.de. My latest
configuration seems to work only when X-lite is running on a PC on my
lan (!!!) and tried to play a call. So I think that there must be some
authentification problem or so...

When x-lite in not running I also get: 403 "Forbidden" ...

sip.conf
--------
...
register => <ACCOUNT-NO>:<SIP-PASSWORD>@sipgate.de

[peer-sipgate]
type=peer
username=<ACCOUNT-NO>
secret=<SIP-PASSWORD>
fromuser=<ACCOUNT-NO>
fromdomain=sipgate.de
host=sipgate.de
context=from-sipgate
...
--------

extension.conf:
---------------
...
exten => _9.,1,Dial(SIP/${EXTEN:1}@peer-sipgate,30,tr)

[from-sipgate]
<calls from sipgate arrive here>
exten => s,1,...
...
---------------

Sascha

-------------------------------------------------------
Sascha Knific           K Systems & Design
Tel. +49-8151-773260    Wittelsbacherstr. 6a
Fax. +49-8151-773262    82319 Starnberg, Germany
Leo  +49-8151-773261    WGS84: N57°59,875' E011°20,568'
knific at k-sysdes.net     http://www.k-sysdes.net


> -----Ursprüngliche Nachricht-----
> Von: asterisk-users-admin at lists.digium.com [mailto:asterisk-users-
> admin at lists.digium.com] Im Auftrag von Birk Bremer
> Gesendet: Freitag, 27. Februar 2004 17:47
> An: asterisk-users at lists.digium.com
> Betreff: [Asterisk-Users] Anybody managed to call a phone through
> sipgate.de
> 
> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
> 
> 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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> Comment: Using GnuPG with Debian - http://enigmail.mozdev.org
> 
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> 
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