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

[David J Carter]

Birk,

Even using VPN to get to the server you will still have I assume a private
IP address on the VPN side. This will pass through a NAT/Firewall to the
outside world. This may or may not be on the server you connect to, but I
would bet you still pass through a NAT/Firewall.

I assume your connection is something like: -

Softphone ---- Asterisk ---- VPN to Server -- Server ---
Firewall/NAT/Router ----- Internet

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 11:32
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] Anybody managed to call a phone through
sipgate.de


-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

The Server I use is somewhere in the Internet with a real ip. Myself and
others connect to the server via vpn in order to go through various
firewalls. Since I can get calls but only can't place calls (via
sipgate.de) I don't think it is a firewall matter...

Birk


David J Carter wrote:
| 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
|
|
| 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
| | |
|
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.2.4 (GNU/Linux)
Comment: Using GnuPG with Debian - http://enigmail.mozdev.org

iD8DBQFAQHwy7QhrwFQeHVsRAgHIAKCcm9fr2CoIVAaTLGLkoUaGF6uZdwCfRaMd
n54rHyhWAMcQSCKXZNTbEfk=
=Mzc2
-----END PGP SIGNATURE-----

_______________________________________________
Asterisk-Users mailing list
Asterisk-Users at lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users