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

[David Hajek]

Is there english version of their sipgate.de website? 

-D 

> -----Original Message-----
> From: asterisk-users-admin at lists.digium.com 
> [mailto:asterisk-users-admin at lists.digium.com] On Behalf Of 
> Birk Bremer
> Sent: Friday, February 27, 2004 7:06 PM
> To: asterisk-users at lists.digium.com
> Subject: Re: [Asterisk-Users] Anybody managed to call a phone 
> through sipgate.de
> 
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> Hi David,
> 
> no the number after the slash is necessary (and yes this is 
> my number) Without that slash/number I'm not able to get a 
> call anymore.
> 
> But thanks
> 
> 	Birk
> 
> 
> 
> 
> David J Carter wrote:
> | 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
> |
> |
> | 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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