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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...

...6200 State:Unavailable Watchers 0 3000 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 -...

...I also haven't really figured out how to 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 -...

...> s,4,GotoIf( [ "${DIALSTATUS}" = "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 Cryptograph...

.../it-stuff/sipfraud/sip-attacker-blacklist And 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 Cryptograph...

...2xxxxx at messages:4] Hangup("Message/ast_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 Cryptograph...

...DP/198.204.241.58/5074",Challenge="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 Cryptograph...

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 *

...just new, so don't expect too much :-). 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

...le but this message repeats continuously: Dropping incompatible voice frame 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 -...

...those: Jan 26 20:06:22 home kernel: [3023015.007826] Bluetooth: hci0 SCO 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 Cryptograph...

...things to do. But... I remember, then, sometimes came over 1000 mails in 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 Cryptograph...

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

I see a lot of attempts by hackers to call 00972595301123? or 011972595115207? or variations but that same 972595 is often present. Can someone break down that dial string with an explanation? The 011 look like an overseas call (from Americas), while the 972595XXXXXX is unclear... -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi list , someone on the list has seen this type of connection attempts in asterisk, fail2ban does not stop 2015-01-08 14:59:47] SECURITY[21515] res_security_log.c: SecurityEvent="ChallengeSent",EventTV="1420750787-386840",Severity="Informational",Service="SIP",EventVersion="1",AccountID="sip:100 at

Hi everyone, How can I introduce some distortion, echo, chopping sound and all other bad quality things that can happen to a SIP trunk? I have plenty of bandwidth and crisp clear lines so the only thing that I can think of is to limit bandwidth but even that requires quite some scripting work. Is there any easy way to simulate a distorted SIP line temporarily for testing? I am appreciate

Hi All; Can anyone advise if using Cisco IP Phones in skinny protocol is fine or not? Or it is better to use it in SIP protocol? Regards Bilal