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: [EMAIL PROTECTED] | [mailto:[EMAIL PROTECTED] Behalf Of Birk Bremer | Sent: 28 February 2004 11:04 | To: [EMAIL PROTECTED] | 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: [EMAIL PROTECTED] | |>[mailto:[EMAIL PROTECTED] On Behalf Of | |>Birk Bremer | |>Sent: Friday, February 27, 2004 7:06 PM | |>To: [EMAIL PROTECTED] | |>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 02115800. | | | | | | dave | | | | | | -Original Message- | | | From: [EMAIL PROTECTED] | | | [mailto:[EMAIL PROTECTED] Behalf Of | | Birk Bremer | | | Sent: 27 February 2004 16:47 | | | To: [EMAIL PROTECTED] | | | 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 => 800:[EMAIL PROTECTED]/02115800 | | | | | | [sipgate] | | | type=friend | | | username=800 | | | secret=SECRET | | | host=sipgate.de | | | fromuser=800 | | | fromdomain=sipgate.net | | | nat=no | | | ;dtmfband=3Dinband | | | context=sipin | | | canreinvite=no | | | | | | | | | extension.conf: | | | exten => _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],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("[EMAIL PROTECTED]/2", | | "SIP/[EMAIL PROTECTED]|30|tr") | | | in new stack | | | ~-- Called [EMAIL PROTECTED] | | | ~-- Got SIP response 403 "Forbidden" back from 217.10.79.9 | | | ~ == No one is available to answer at this time | | | ~-- Hungup '[EMAIL PROTECTED]/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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anybody managed to call a phone through sipgate.de
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 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: [EMAIL PROTECTED] |>[mailto:[EMAIL PROTECTED] On Behalf Of |>Birk Bremer |>Sent: Friday, February 27, 2004 7:06 PM |>To: [EMAIL PROTECTED] |>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 02115800. | | | | dave | | | | -Original Message- | | From: [EMAIL PROTECTED] | | [mailto:[EMAIL PROTECTED] Behalf Of | Birk Bremer | | Sent: 27 February 2004 16:47 | | To: [EMAIL PROTECTED] | | 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 => 800:[EMAIL PROTECTED]/02115800 | | | | [sipgate] | | type=friend | | username=800 | | secret=SECRET | | host=sipgate.de | | fromuser=800 | | fromdomain=sipgate.net | | nat=no | | ;dtmfband=3Dinband | | context=sipin | | canreinvite=no | | | | | | extension.conf: | | exten => _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],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("[EMAIL PROTECTED]/2", | "SIP/[EMAIL PROTECTED]|30|tr") | | in new stack | | ~-- Called [EMAIL PROTECTED] | | ~-- Got SIP response 403 "Forbidden" back from 217.10.79.9 | | ~ == No one is available to answer at this time | | ~-- Hungup '[EMAIL PROTECTED]/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 => 800:[EMAIL PROTECTED]/800 | | ^ can't be correct | | | | | | | | | | | | [sipgate] | | | | | | type=friend | | | | | | username=800 | | | | | | secret=sipgatepasswort | | | | | | host=sipgate.de | | | | | | fromuser=800 | | | | | | fromdomain=sipgate.net | | | | | | nat=yes | | | | | | ;dtmfband=inband | | | | | | context=incomingsipgate | | | | | | canreinvite=no | | | | | | | | | | | | Aus der extensions.conf : | | | | | | | | | | | | [incomingsipgate] | | | | | | exten => h,1,Hangup | | | | | | exten => 800,1,Dial(SIP/internestelefon,20,tr) | | | | | | | | | | | | [sipgate] | | | | | | exten => _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,tr) | | | | | | exten => _9.,2,Playback(invalid) | | | | | | exten => _9.,3,Hangup | | ___ | Asterisk-Users mailing list | [EMAIL PROTECTED] | http://lists.digium.com/mailman/listinfo/asterisk-users | To UNSUBSCRIBE or update options visit: | ~ http://lists.digium.com/mailman/listinfo/asterisk-users | | ___ | Asterisk-Users mailing list | [EMAIL PROTECTED] | http://lists.digium.com/mailman/listinfo/asterisk-users | To UNSUBSCRIBE or update options visit: | ~ http://lists.digium.com/mailman/listinfo/asterisk-users | ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: ~ http://lists.digium.com/mailman/listinfo/asterisk-users | ___ | Asterisk-Users mailing list | [EMAIL PROTECTED] | http://lists.digium.com/mailman/listinfo/asterisk-users | To UNSUBSCRIBE or update options visit: |http://lists.digium.com/mailman/listinfo/asterisk-users -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) Comment: Using GnuPG with Debian - http://enigmail.mozdev.org iD8D
Re: [Asterisk-Users] Anybody managed to call a phone through sipgate.de
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hello Philipp, whis also did not help - still a: - -- Got SIP response 403 "Forbidden" back from 217.10.79.9 But thanks (do you have working configuration?) Birk Philipp von Klitzing wrote: | Hi! | | |>has anybody managed to call a (old fashioned) phone using Sipgate.de and |>asterisk? (yes I have money on my account :-) ) |> |>extension.conf: |>exten => _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,tr) | | | Try this instead: | exten => _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,tr) | | Philipp | | | ___ | Asterisk-Users mailing list | [EMAIL PROTECTED] | http://lists.digium.com/mailman/listinfo/asterisk-users | To UNSUBSCRIBE or update options visit: |http://lists.digium.com/mailman/listinfo/asterisk-users -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) Comment: Using GnuPG with Debian - http://enigmail.mozdev.org iD8DBQFAP4sS7QhrwFQeHVsRAjkrAKCmh2XkOGhm7frAh4dtgCGN55C5wACdEYSo S4DBVGM58t4C9UjU4i/LylA= =K4J5 -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anybody managed to call a phone through sipgate.de
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi David, no the number after the slash is necessary (and yes this is my number) Without that slash/number I'm not able to get a call anymore. But thanks Birk David J Carter wrote: | Hi, | | I would be tempted to get rid of the slash and number on the register line, | unless your asterisk extension is 02115800. | | dave | | -Original Message- | From: [EMAIL PROTECTED] | [mailto:[EMAIL PROTECTED] Behalf Of Birk Bremer | Sent: 27 February 2004 16:47 | To: [EMAIL PROTECTED] | Subject: [Asterisk-Users] Anybody managed to call a phone through | sipgate.de | | | Hello everybody, | | has anybody managed to call a (old fashioned) phone using Sipgate.de and | asterisk? (yes I have money on my account :-) ) | | | The configuration I got from the sipgate.de people is at the botton of | the mail | | | Here is mine: | | sip.conf: | | register => 800:[EMAIL PROTECTED]/02115800 | | [sipgate] | type=friend | username=800 | secret=SECRET | host=sipgate.de | fromuser=800 | fromdomain=sipgate.net | nat=no | ;dtmfband=3Dinband | context=sipin | canreinvite=no | | | extension.conf: | exten => _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],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("[EMAIL PROTECTED]/2", "SIP/[EMAIL PROTECTED]|30|tr") | in new stack | ~-- Called [EMAIL PROTECTED] | ~-- Got SIP response 403 "Forbidden" back from 217.10.79.9 | ~ == No one is available to answer at this time | ~-- Hungup '[EMAIL PROTECTED]/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 => 800:[EMAIL PROTECTED]/800 | ^ can't be correct | | | | | | | | [sipgate] | | | | type=friend | | | | username=800 | | | | secret=sipgatepasswort | | | | host=sipgate.de | | | | fromuser=800 | | | | fromdomain=sipgate.net | | | | nat=yes | | | | ;dtmfband=inband | | | | context=incomingsipgate | | | | canreinvite=no | | | | | | | | Aus der extensions.conf : | | | | | | | | [incomingsipgate] | | | | exten => h,1,Hangup | | | | exten => 800,1,Dial(SIP/internestelefon,20,tr) | | | | | | | | [sipgate] | | | | exten => _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,tr) | | | | exten => _9.,2,Playback(invalid) | | | | exten => _9.,3,Hangup ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: ~ http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: ~ http://lists.digium.com/mailman/listinfo/asterisk-users -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) Comment: Using GnuPG with Debian - http://enigmail.mozdev.org iD8DBQFAP4b07QhrwFQeHVsRAvokAJ9flLxgaKalQH7Qjlro/sJBweu/LwCeO//S gtjYXR78PiVK9xRbZnb6Oqs= =nnhy -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Anybody managed to call a phone through sipgate.de
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hello everybody, has anybody managed to call a (old fashioned) phone using Sipgate.de and asterisk? (yes I have money on my account :-) ) The configuration I got from the sipgate.de people is at the botton of the mail Here is mine: sip.conf: register => 800:[EMAIL PROTECTED]/02115800 [sipgate] type=friend username=800 secret=SECRET host=sipgate.de fromuser=800 fromdomain=sipgate.net nat=no ;dtmfband=3Dinband context=sipin canreinvite=no extension.conf: exten => _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],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("[EMAIL PROTECTED]/2", "SIP/[EMAIL PROTECTED]|30|tr") in new stack ~-- Called [EMAIL PROTECTED] ~-- Got SIP response 403 "Forbidden" back from 217.10.79.9 ~ == No one is available to answer at this time ~-- Hungup '[EMAIL PROTECTED]/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 => 800:[EMAIL PROTECTED]/800 ^ can't be correct | | | | [sipgate] | | type=friend | | username=800 | | secret=sipgatepasswort | | host=sipgate.de | | fromuser=800 | | fromdomain=sipgate.net | | nat=yes | | ;dtmfband=inband | | context=incomingsipgate | | canreinvite=no | | | | Aus der extensions.conf : | | | | [incomingsipgate] | | exten => h,1,Hangup | | exten => 800,1,Dial(SIP/internestelefon,20,tr) | | | | [sipgate] | | exten => _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] running asterisk as non-root
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hello everyone Due to security reasons I want to run asterisk as a non root. I normaly installed asterisk, created an * user, moved the binaries to /usr/bin and chowned all the files and directories mentiont in the * manual (handbook-draft.pdf) Now I can start * but I get the following warning (which I don't get if I run it as a root): Feb 14 19:10:53 WARNING[213006]: pbx_wilcalu.c:69 autodial: Autodial: Unable to open file ~ == Parsing '/etc/asterisk/enum.conf': Found I don't know if * really works - I have't tired jet - can anybody tell me which file * want's to access? ( I looked in the source but I'm not that familiar with the code) Thank's in advance Birk Bremer -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) Comment: Using GnuPG with Debian - http://enigmail.mozdev.org iD8DBQFALoxP7QhrwFQeHVsRAq/ZAJ0VE5pGY98Ip+FlbvPYv4bHOEoXXACgkYSK m8hpZA/orrMBMRb4NoKLoJk= =7BH7 -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Can Asterisk act like a normal sip phone?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hello everyone, I'm relatively new to the subject - so pleace don't punish me for idiotic questions. ;-) Can Asterisk act like a normal Sip phone and e.g. connect to another sip-gateway? Background: There is a new german company at: http://www.sipgate.de (sorry German only page) They offer a a gateway between a real telephone number and their sip server. (at the moment for free) If you had the possibility to connect asterisk as a phone to this server it would be an easy (and cheap!) way to realise a gateway to old-style-phoneline. Waiting for reply, Birk Bremer -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.3 (GNU/Linux) Comment: Using GnuPG with Debian - http://enigmail.mozdev.org iD8DBQFAGg6+7QhrwFQeHVsRAo+1AJ9+gk79nIxbxt6rPPpHIBw2MZibBQCdEcJN wWawRjIjmpUs9orqrmEEcNI= =EGhT -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users