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Birk Bremer

2004-Feb-27 09:46 UTC

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

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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@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@myself]/2",
"SIP/number@sipgate.de|30|tr")
in new stack
~    -- Called number@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@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@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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David J Carter

2004-Feb-27 10:14 UTC

head link

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

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@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com]On Behalf Of Birk Bremer
Sent: 27 February 2004 16:47
To: asterisk-users@lists.digium.com
Subject: [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@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@myself]/2",
"SIP/number@sipgate.de|30|tr")
in new stack
~    -- Called number@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@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@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
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.2.4 (GNU/Linux)
Comment: Using GnuPG with Debian - http://enigmail.mozdev.org

iD8DBQFAP3R87QhrwFQeHVsRAjx+AJ9SvPdV4YY5iSZflo9XX/Xi97YM3wCghniD
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http://lists.digium.com/mailman/listinfo/asterisk-users
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   http://lists.digium.com/mailman/listinfo/asterisk-users

Sascha Knific

2004-Feb-27 10:45 UTC

head link

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

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@k-sysdes.net     http://www.k-sysdes.net

> -----Urspr?ngliche Nachricht-----
> Von: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-
> admin@lists.digium.com] Im Auftrag von Birk Bremer
> Gesendet: Freitag, 27. Februar 2004 17:47
> An: asterisk-users@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@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/${EXTEN:1}@sipgate.de,30,tr)
> 
> To be called on my sipgate number - no problem
> 
> If I want to call somebody I get the following error:
> 
> When I call a number directly out of the softphone:
> Executing Dial("IAX2[myself@myself]/2",
"SIP/number@sipgate.de|30|tr")
> in new stack
> ~    -- Called number@sipgate.de
> ~    -- Got SIP response 403 "Forbidden" back from 217.10.79.9
> ~  == No one is available to answer at this time
> ~    -- Hungup 'IAX2[myself@myself]/2
> 
> 
> 
> when I use the webinterface at sipgate.de I get a ring at my
softphone,> when I pick the call I get the message (in the appearing box)
> "Teilnehmer nicht gefunden" - User/Number not found
> 
> sometimes (while tried different config. I also got (at * console) to
> many hops...
> 
> 
> Has anybody managed this - can you please send me your configuration
> (sip, extensions) .... or can anybody help
> 
> Thanks in advance
> 
> 		Birk Bremer
> 
> 
> 
> 
> 
> The configuration the sipgate people suggest:
> 
> ~ > register => 800XXXX:sipgatepasswort@sipgate.de/800XXXX
> 						  ^^^^^ can't be correct
> |
> |
> |
> | [sipgate]
> |
> | type=friend
> |
> | username=800XXXX
> |
> | secret=sipgatepasswort
> |
> | host=sipgate.de
> |
> | fromuser=800XXXX
> |
> | fromdomain=sipgate.net
> |
> | nat=yes
> |
> | ;dtmfband=inband
> |
> | context=incomingsipgate
> |
> | canreinvite=no
> |
> |
> |
> | Aus der extensions.conf :
> |
> |
> |
> | [incomingsipgate]
> |
> | exten => h,1,Hangup
> |
> | exten => 800XXXX,1,Dial(SIP/internestelefon,20,tr)
> |
> |
> |
> | [sipgate]
> |
> | exten => _9.,1,Dial(SIP/${EXTEN:1}@sipgate,30,tr)
> |
> | exten => _9.,2,Playback(invalid)
> |
> | exten => _9.,3,Hangup
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v1.2.4 (GNU/Linux)
> Comment: Using GnuPG with Debian - http://enigmail.mozdev.org
> 
> iD8DBQFAP3R87QhrwFQeHVsRAjx+AJ9SvPdV4YY5iSZflo9XX/Xi97YM3wCghniD
> 5HUMSd5i2HUik75eajuJtpU> =01sy
> -----END PGP SIGNATURE-----
> 
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@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

Philipp von Klitzing

2004-Feb-27 11:03 UTC

head link

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

Hi!
> has anybody managed to call a (old fashioned) phone using Sipgate.de and
> asterisk? (yes I have money on my account :-) )
>
> extension.conf:
> exten => _9.,1,Dial(SIP/${EXTEN:1}@sipgate.de,30,tr)
Try this instead:
exten => _9.,1,Dial(SIP/${EXTEN:1}@sipgate,30,tr)

Philipp

Birk Bremer

2004-Feb-27 11:23 UTC

head link

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

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Hash: SHA1

Hello Philipp,

whis also did not help - still a:

- -- Got SIP response 403 "Forbidden" back from 217.10.79.9

But thanks (do you have working configuration?)

Birk



Philipp von Klitzing wrote:
| Hi!
|
|
|>has anybody managed to call a (old fashioned) phone using Sipgate.de and
|>asterisk? (yes I have money on my account :-) )
|>
|>extension.conf:
|>exten => _9.,1,Dial(SIP/${EXTEN:1}@sipgate.de,30,tr)
|
|
| Try this instead:
| exten => _9.,1,Dial(SIP/${EXTEN:1}@sipgate,30,tr)
|
| Philipp
|
|
| _______________________________________________
| Asterisk-Users mailing list
| Asterisk-Users@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

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asterisk users - Feb 2004 - Anybody managed to call a phone through sipgate.de

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

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

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

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

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