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...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-----

...alk_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-----

...risk -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

...king/locking.c:(174) ERROR: Failed to initialise share modes We use Samba 2.0.5 on an SCO Release 5.0.5, Kernel rel=3.2 128Mo RAM Can you give me some orientation in my problem ? Is it a memory space problem or something like this ? Thanks for your help -- Francois Heckler Transiciel Regions Lorraine 03.83.15.22.26 francois.heckler@est.transiciel.com --

....." 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-----

...n => 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/pipermail/asterisk-users/attac...

...L: 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/pipermail/asterisk-users/attac...

...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

...ot;) 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/pipermail/asterisk-users/attac...

...gt; 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/pipermail/asterisk-users/attac...

Fellow hecklers, Just spent a while getting this working. Turns out heckle will heckle a whole module and sub-modules with one call, so with a bit of string matching, you can build a nice tool to heckle your whole app and report any failures. Was going to post to the list, as the first version was about...

...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 > To UNSUBSCRIBE or update options visit: > http...

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 *

...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

...ame on Mobile/DNA-54f4 of format slin since our native format has changed to 0x0 (nothing) Can somebody point me to the right direction? Asterisk SVN-branch-1.6.2-r313579 - -Stefan - -- (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) iEYEARECAAYFAk2tr3YACgkQbQKZlCdPOMOLkwCfYm/jdPx3uOYdcvZ5XsZeKWAg sD8AoL4ygna6jWsKLY9sEwzU2VjRek/T =hbxJ -----END PGP SIGNATURE-----

...packet for unknown connection handle 65328 Mobile integration is the last bit that my * is missing before being the perfect PBX, so I hope, somebody here could help me with that :). -- (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/pipermail/asterisk-users/attac...

...24h. Now it's hardly 50 new mails per week. Is the list dead? Or is the project dead? Or is nobody tinkering any more and everybody buying some turnkey-stuff? Just wondering... -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/pipermail/asterisk-users/attac...

Hi, Does anyone know how to make this thing (7940) work with asterisk (chan_sccp module) ? I've set the configuration according to the wiki and now the phone just keep asking for CTLSEP<xxx>.tlv from my tftp server. In the cisco's web interface, I found this in the device logs : 0x8106, 0x0, 0x12300800 0x8106, 0x0, 0x12300800 0x8106, 0x0, 0x12300800 0x8106, 0x0, 0x12300800 ...

I am looking for a simple way to forward calls unconditionally with Asterisk. We are running an Asterisk system with 10 extensions using SIP. One of our users leaves the office regulary, when she is out, she needs to be able to forward unconditionally to her mobile or collegue. I am trying to keep it as simple as possible, we use Cisco 7940's, they have a call forward option, when she