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

David J Carter david.carter at codepipe.com
Sat Feb 28 04:18:06 MST 2004


Hi,

Are you behind a NAT/Firewall?

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: 28 February 2004 11:04
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] Anybody managed to call a phone through
sipgate.de


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David Hajek wrote:
| Is there english version of their sipgate.de website?


no ... I just tried the google translater - it did not work (for me) I
think the translation programs don't work with php pages...

Birk


|
| -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
|>
| 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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