Birk Bremer
2004-Feb-27 09:46 UTC
[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-----
David J Carter
2004-Feb-27 10:14 UTC
[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 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
Sascha Knific
2004-Feb-27 10:45 UTC
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/${EXTEN:1}@peer-sipgate,30,tr) [from-sipgate] <calls from sipgate arrive here> exten => s,1,... ... --------------- Sascha ------------------------------------------------------- Sascha Knific K Systems & Design Tel. +49-8151-773260 Wittelsbacherstr. 6a Fax. +49-8151-773262 82319 Starnberg, Germany Leo +49-8151-773261 WGS84: N57?59,875' E011?20,568' knific@k-sysdes.net http://www.k-sysdes.net> -----Urspr?ngliche Nachricht----- > Von: asterisk-users-admin@lists.digium.com [mailto:asterisk-users- > admin@lists.digium.com] Im Auftrag von Birk Bremer > Gesendet: Freitag, 27. Februar 2004 17:47 > An: asterisk-users@lists.digium.com > Betreff: [Asterisk-Users] Anybody managed to call a phone through > sipgate.de > > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > Hello everybody, > > has anybody managed to call a (old fashioned) phone using Sipgate.deand> asterisk? (yes I have money on my account :-) ) > > > The configuration I got from the sipgate.de people is at the botton of > the mail > > > Here is mine: > > sip.conf: > > register => 800XXXX:SECRET@sipgate.de/02115800XXXX > > [sipgate] > type=friend > username=800XXXX > secret=SECRET > host=sipgate.de > fromuser=800XXXX > fromdomain=sipgate.net > nat=no > ;dtmfband=3Dinband > context=sipin > canreinvite=no > > > extension.conf: > exten => _9.,1,Dial(SIP/${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 mysoftphone,> 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/${EXTEN:1}@sipgate,30,tr) > | > | exten => _9.,2,Playback(invalid) > | > | exten => _9.,3,Hangup > -----BEGIN PGP SIGNATURE----- > Version: GnuPG v1.2.4 (GNU/Linux) > Comment: Using GnuPG with Debian - http://enigmail.mozdev.org > > iD8DBQFAP3R87QhrwFQeHVsRAjx+AJ9SvPdV4YY5iSZflo9XX/Xi97YM3wCghniD > 5HUMSd5i2HUik75eajuJtpU> =01sy > -----END PGP SIGNATURE----- > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users
Philipp von Klitzing
2004-Feb-27 11:03 UTC
[Asterisk-Users] Anybody managed to call a phone through sipgate.de
Hi!> has anybody managed to call a (old fashioned) phone using Sipgate.de and > asterisk? (yes I have money on my account :-) ) > > extension.conf: > exten => _9.,1,Dial(SIP/${EXTEN:1}@sipgate.de,30,tr)Try this instead: exten => _9.,1,Dial(SIP/${EXTEN:1}@sipgate,30,tr) Philipp
Birk Bremer
2004-Feb-27 11:23 UTC
[Asterisk-Users] Anybody managed to call a phone through sipgate.de
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello Philipp, whis also did not help - still a: - -- Got SIP response 403 "Forbidden" back from 217.10.79.9 But thanks (do you have working configuration?) Birk Philipp von Klitzing wrote: | Hi! | | |>has anybody managed to call a (old fashioned) phone using Sipgate.de and |>asterisk? (yes I have money on my account :-) ) |> |>extension.conf: |>exten => _9.,1,Dial(SIP/${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 -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.2.4 (GNU/Linux) Comment: Using GnuPG with Debian - http://enigmail.mozdev.org iD8DBQFAP4sS7QhrwFQeHVsRAjkrAKCmh2XkOGhm7frAh4dtgCGN55C5wACdEYSo S4DBVGM58t4C9UjU4i/LylA=K4J5 -----END PGP SIGNATURE-----