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

[Birk Bremer]

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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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