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-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, has anyone managed to establish an XMPP connection to the facebook Jabber servers? I'd like to send messages on missed calls vie FB. - -S - -- (o_ Stefan Gofferje | SCLT, MCP, CCSA //\ Reg'd Linux User #247167 | VCP #2263 V_/_ Heckler & Koch - the original point and click interface -----BEGIN PGP SIGNATURE----- Version: GnuPG v2.0.16 (GNU/Linux) iEYEARECAAYFAk2qIpsACgkQbQKZlCdPOMOEcACfcVot6VqUDB/99PNXT2C+Bv5l QBwAnAr4yQjIg03IcwOHg4hnSCv5LrLT =Y112 -----END PGP SIGNATURE-----

...000 at internal : gtalk/gtalk_account/ State:Idle Watchers 1 Funnily, the gtalk hint is the only one with a watcher although all hints are hooked in various phones... Any ideas, comments, etc...? - -S - -- (o_ Stefan Gofferje | SCLT, MCP, CCSA //\ Reg'd Linux User #247167 | VCP #2263 V_/_ Heckler & Koch - the original point and click interface -----BEGIN PGP SIGNATURE----- Version: GnuPG v2.0.16 (GNU/Linux) iEYEARECAAYFAk2qJgYACgkQbQKZlCdPOMN1xgCaA1Fk82FXF41AdImMU358VzDy kfUAoIkDK7qCx2Xjwn2bd/osg1rvuqBP =Z8lO -----END PGP SIGNATURE-----

G'Day All, Can someone help me out please. My new CISCO 7960's manual says I have to setup a TFTP server. Googled it and got a little understanding, but from * standpoint, well I am still a lost. Can I set this tftp server on the same * box? Can in be on a WinXP box? Which tftp software would you recommend? Thanks much. BTY: Does anyone have a How-To on getting the 7960 fully configured

...ect to a already running asterisk with asterisk -R, it's never colored, despite I'm running both from the same console (tty). Is there a way to force asterisk -R into color mode? Regards, Stefan -- (o_ Stefan Gofferje | Linux Systems Specialist //\ Reg'd Linux User #247167 | Network Security Specialist V_/_ Heckler & Koch - the original point and click interface

...create the "stefan at .." solution. To reach this context, people have to call IAX/guest at my.asterisk.pbx How do I create a context in which all calls from nonregistered clients are handled? - -S - -- (o_ Stefan Gofferje | SCLT, MCP, CCSA //\ Reg'd Linux User #247167 | VCP #2263 V_/_ Heckler & Koch - the original point and click interface -----BEGIN PGP SIGNATURE----- Version: GnuPG v2.0.16 (GNU/Linux) iEYEARECAAYFAk3xBXYACgkQbQKZlCdPOMN6qgCfR+TBVpVCSKDZyzUJk6r53VYS dvYAoJrbb76zqFY3c1K0YzA9j3dowPE4 =2SCj -----END PGP SIGNATURE-----

...= "BUSY" ]?2) exten => s,5,Hangup For that I'd need to somewhere get the last dialed number from the channel/line I'm initiating the call from. Is something like this already implemented? -S -- (o_ Stefan Gofferje | SCLT, MCP, CCSA //\ Reg'd Linux User #247167 | VCP #2263 V_/_ Heckler & Koch - the original point and click interface -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/pkcs7-signature Size: 4079 bytes Desc: S/MIME Cryptographic Signature URL: <http://lists.digium.com/p...

...d that's the download URL: http://stefan.gofferje.net/sipblocklist.zone Note that the list is updated every 6h so polling it more often doesn't help anything. Please limit polling to once a day or so. -S -- (o_ Stefan Gofferje | SCLT, MCP, CCSA //\ Reg'd Linux User #247167 | VCP #2263 V_/_ Heckler & Koch - the original point and click interface -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/pkcs7-signature Size: 4079 bytes Desc: S/MIME Cryptographic Signature URL: <http://lists.digium.com/p...

...nt. The a() option of the dial app only works for CALLED parties and when trying to use a macro with the m() option, the Playback also goes to the called party. Anyone any hints on that? Regards, Stefan -- (o_ Stefan Gofferje | Linux Systems Specialist //\ Reg'd Linux User #247167 | Network Security Specialist V_/_ Heckler & Koch - the original point and click interface

...pg123 in my tasklist, consuming 97% of CPU power. That > seems a little too much for me, considering, I have just 4 > external lines... > > Regards, > Stefan > > > -- > (o_ Stefan Gofferje | Linux Systems Specialist > //\ Reg'd Linux User #247167 | SuSE Certified Linux Trainer > V_/_ Linux is like a Wigwam - No gates, no windows, Apache inside > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users >...

Hello, list a have a problem i can start asterisk, i get the fowlling error: [chan_capi.so] => (Common ISDN API for Asterisk) == Parsing '/etc/asterisk/capi.conf': Found Feb 11 13:50:36 NOTICE[2535]: chan_capi.c:2636 load_module: CAPI not installed! Feb 11 13:50:36 WARNING[2535]: loader.c:345 ast_load_resource: chan_capi.so: load_module failed, returning -1 Feb 11 13:50:36

...t_msg_queue", "") in new stack == Spawn extension (messages, +3584172xxxxx, 4) exited non-zero on 'Message/ast_msg_queue' Is there a way to make * ignore the comma in the string here? -S -- (o_ Stefan Gofferje | SCLT, MCP, CCSA //\ Reg'd Linux User #247167 | VCP #2263 V_/_ Heckler & Koch - the original point and click interface -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/pkcs7-signature Size: 4079 bytes Desc: S/MIME Cryptographic Signature URL: <http://lists.digium.com/p...

...23965594" > > > I modified the fail2ban with the filter, but still not detected Do you really want to detect "ChallengeSent"? That should occur also on legitimate login processes... -S -- (o_ Stefan Gofferje | SCLT, MCP, CCSA //\ Reg'd Linux User #247167 | VCP #2263 V_/_ Heckler & Koch - the original point and click interface -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/pkcs7-signature Size: 4079 bytes Desc: S/MIME Cryptographic Signature URL: <http://lists.digium.com/p...

...l(b6000) > > Unfortunately, this doesn't work... Would probably have bee too easy... > > Does anybody have a clue on how to complete this? > > Regards, > Stefan > > -- > (o_ Stefan Gofferje | Linux Systems Specialist > //\ Reg'd Linux User #247167 | SuSE Certified Linux Trainer > V_/_ Linux is like a Wigwam - No gates, no windows, Apache inside > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > T...

..._sipgate or both together at a > time", leading any more caller from inside or outside to receive a busy > or congestion signalisation? > > Regards, > Stefan > > > -- > (o_ Stefan Gofferje | Linux Systems Specialist > //\ Reg'd Linux User #247167 | SuSE Certified Linux Trainer > V_/_ Linux is like a Wigwam - No gates, no windows, Apache inside > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users >...

...en dropped. Is this because of ethical reasons or because it was too buggy and not fixable? Is there something similar? Or is there an "unofficial" source where to get it? Regards, Stefan -- (o_ Stefan Gofferje | Linux Systems Specialist //\ Reg'd Linux User #247167 | SuSE Certified Linux Trainer V_/_ Linux is like a Wigwam - No gates, no windows, Apache inside _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options v...

...sterisk quits without further notice > and log-entry. When 8004 is dialled from any other source, > everything is fine. > > Any clues? > > Regards, > Stefan > > -- > (o_ Stefan Gofferje | Linux Systems Specialist > //\ Reg'd Linux User #247167 | SuSE Certified Linux Trainer > V_/_ Linux is like a Wigwam - No gates, no windows, Apache inside > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users >...

...SIP version of 7960. The SCCP image seems to > ignore that. Is there any way to have asterisk taking > influence on the > phone's ringing? > > Regards, > Stefan > > -- > (o_ Stefan Gofferje | Linux Systems Specialist > //\ Reg'd Linux User #247167 | Network Security SpecialistHi folks > V_/_ Heckler & Koch - the original point and click interface > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users...

Hi, I tried to connect cisco 7910 into asterisk system using chan_sccp.so. But I got a major issue : - when I called from 7910 to another sip phone in the same asterisk server, the call took place normally. - when I called from 7910 to another sip phone in different asterisk server, the call is answered but I cannot hear nor say anything. The phone just immediately lose its tone. - when I got

hi PSTN <--> [Teles ISDN / Asterisk] <--> SIP client When call is made through ISDN, no matter if taken from PSTN or Asterisk side, person in PSTN side can hear perfectly but in Asterisk side I only hear a very scrambled or very low quality voice, words repeated several times. Same is with echo test (call taken from PSTN) Setup: * Teles 16.3 ISA ISDN card with hisax kernel module *

...uch :-). Everybody is invited to participate especially at the forum. Any comments, proposals, critics are very welcome. Find the site at http://chan-sccp.org/ and the forum at http://forum.chan-sccp.org/ Regards, Stefan -- (o_ Stefan Gofferje | SCLT //\ Reg'd Linux User #247167 | VCP #2263 V_/_ Heckler & Koch - the original point and click interface