Re: [asterisk-users] Asterisk 1.8.0 Release Candidate 2 Now Available
Hello All, To help out with dependency issues, a wiki at http://asteriskdependencies.linuxinnovations.com was set up. It's fairly new, so any contributors interested would need to fill it with the proper data. -Elliot On Mon, Sep 27, 2010 at 10:15 PM, Leif Madsen wrote: > On 10-09-26 02:55 PM, Ira wrote: >> At 10:37 PM 9/24/2010, you wrote: >>> You probably need to install libssl-dev then rerun ./configure. At >>> least I did (Debian Lenny). Seems chan_sip needs res_crypto which >>> needs libssl. >> >> Thanks, I tried to figure out what I needed but I failed. That was >> it, though on CentOS it seems to be openssl-devel. > > FYI, this is no longer an issue as of today. I opened an issue per the > Asterisk > development team, and Tilghman fixed the issue. > > https://issues.asterisk.org/view.php?id=18062 > > The next release candidate will allow chan_sip to use, but not require, the > OpenSSL development libraries. > > Thanks! > Leif. > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP flood attacK
On Sun, Oct 03, 2010 at 02:19:35PM -0600, Greg Saunders wrote: > Hello all. I was recently the victim of a SIP flood attack. I'm wondering > what is the best method to prevent such things in the future. In sip.conf: [general] alwaysauthreject = yes The attacking program is probably svwar.py (part of SIPVicious). It will give up as soon as it realizes it can't tell the difference between attempting to register an invalid extension and a valid one (with an arbitrary password). It's the default in 1.8, but the option goes back at least to 1.4. -- Barry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP flood attacK
I have a decent thread on Team Forrest about this. Also, good to read up on John Todd's 7 deadly sins. http://www.teamforrest.com/blog/171/asterisk-no-matching-peer-found-block/ ---fred http://qxork.com On Oct 3, 2010, at 4:51 PM, Alec Davis wrote: > Make sure you have allowguest=no in your sip.conf, the default is yes, unless > you really do want anonymous guests. > > Also it might pay to consider > http://www.emergingthreats.net/index.php/rules-mainmenu-38.html > > Alec Davis > > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Greg Saunders > Sent: Monday, 4 October 2010 9:20 a.m. > To: asterisk-users@lists.digium.com > Subject: [asterisk-users] SIP flood attacK > > Hello all. I was recently the victim of a SIP flood attack. I'm wondering > what is the best method to prevent such things in the future. > Many thanks > Greg > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Attempts to hack Asterisk - What do these lines means
In another email I've just responded to, it might pay to consider http://www.emergingthreats.net/index.php/rules-mainmenu-38.html Alec Davis _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bruce bruce Sent: Sunday, 3 October 2010 7:59 a.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Attempts to hack Asterisk - What do these lines means Hi Everyone, Like always, here are IPs from China that try to hack an Asterisk server. Can someone please explain what is happening or what the hacker is trying to reach: 02/10/2010 11:10 SIP/113.105.152.51-00fb sip "sip" s ANSWERED 13 02/10/2010 11:10 SIP/113.105.152.51-00fe sip "sip" s ANSWERED 13 02/10/2010 11:10 SIP/113.105.152.51-00fc sip "sip" s ANSWERED 13 02/10/2010 11:10 SIP/113.105.152.51-00fd sip "sip" s ANSWERED 13 02/10/2010 11:10 SIP/113.105.152.51-00ff sip "sip" s ANSWERED 13 02/10/2010 11:10 SIP/113.105.152.51-0100 sip "sip" s ANSWERED 13 02/10/2010 11:17 SIP/222.73.204.198-0101 sip "sip" s ANSWERED 13 02/10/2010 11:17 SIP/222.73.204.198-0102 sip "sip" s ANSWERED 13 02/10/2010 11:17 SIP/222.73.204.198-0103 sip "sip" s ANSWERED 13 02/10/2010 11:17 SIP/222.73.204.198-0104 sip "sip" s ANSWERED 13 02/10/2010 11:17 SIP/222.73.204.198-0105 sip "sip" s ANSWERED 13 02/10/2010 11:17 SIP/222.73.204.198-0106 sip "sip" s ANSWERED 13 02/10/2010 11:17 SIP/222.73.204.198-0107 sip "sip" s ANSWERED 13 02/10/2010 11:17 SIP/222.73.204.198-0108 sip "sip" s ANSWERED 13 02/10/2010 11:17 SIP/222.73.204.198-0109 sip "sip" s ANSWERED 13 Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP flood attacK
Make sure you have allowguest=no in your sip.conf, the default is yes, unless you really do want anonymous guests. Also it might pay to consider http://www.emergingthreats.net/index.php/rules-mainmenu-38.html Alec Davis _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Greg Saunders Sent: Monday, 4 October 2010 9:20 a.m. To: asterisk-users@lists.digium.com Subject: [asterisk-users] SIP flood attacK Hello all. I was recently the victim of a SIP flood attack. I'm wondering what is the best method to prevent such things in the future. Many thanks Greg -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP flood attacK
Hello all. I was recently the victim of a SIP flood attack. I'm wondering what is the best method to prevent such things in the future. Many thanks Greg -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] more condition check for gotoif
Hello, is it possible to check more than one condition for GOTOIF in the dialplan? Or is the normal way to cascade the diaplan each GOTOIF? The Background is that I would like to check more than 2 values from a Variable, and then route the call based on the value. Thanks for your help. Daniel Knoll -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Security - Using Linksys PAP2T from outside with a dynamic IP is there anyway to block all other traffic but those of the PAP2T?
Thanks for the input guys. So, the IP is resolved only when IPTABLES is loaded or reloaded. Therefore, the best approach would be to ping the hostname every let's say 3 seconds and see if the IP is still the same and if it is then move on, otherwise update the iptables with the new IP address. This sounds it would work but I am not sure how fast DynDns can resolve the IP for me (delay) and I am looking to connect 40 PAP2T to this system. So, all in all that is 40 queries to DynDNS each 3 seconds. As I mentioned earlier, wouldn't it be more solid if I run my own Dynamic DNS server on the same box as Asterisk (is that even possible?) and what sort of other security holes would I be exposing doing that? Thanks again for all the great input. -Bruce On Sun, Oct 3, 2010 at 8:01 AM, Steve Edwards wrote: > On Sat, 2 Oct 2010, Kyle Kienapfel wrote: > > > You're not going to be able to put a dns hostname in the iptables, but > > you could have a script that runs at times and gets the ip address for > > your dynamic hostname and allows that. > > Almost. > > You can put a host name in iptables, but it is resolved when loaded. > > You could restart iptables when your dynamic host name changes and it will > be resolved correctly with the new IP address. > > -- > Thanks in advance, > - > Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST > Newline Fax: +1-760-731-3000 > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] debian/dahdi/zaphfc - Unable to receive TEI fromnetwork!
Am 02.10.2010 15:55, schrieb Alex: > I'm not that much thrilled by ISDN these days, I mostly want to get back > to a working setup. But since I've battled with this for a few days, if > a few more days are needed to help debug what appears to be a problem > with vzaphfc (?), I can spend some time. That is, if you care to provide > test scenarios and/or test/instrumented code. Tell me if this needs to > be moved off (this) list... jabber would be fine, if you say so. Exactly the same scenario for me. I talked to tzafrir on IRC but unfortunately I did that during a very busy time for me so I wasn't able to devote the time it would have needed. For me problems started when upgrading to DAHDI 2.4.0 (from 2.3.0.1). But maybe the fault in my case lies in libpri 1.4.12 because that is what's needed for Asterisk 1.8.0. However, downgrading to 2.3.0.1 (even with the newer libpri) makes things work again. So I'm almost sure that it's somewhere in the DAHDI/vzaphfc code. Felix -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Flash WAV Player
> Does anyone know of a Flash or Java player that can play WAV files created by > Asterisk? Found one. http://blog.datacompboy.ru/2010/01/27/wavplayer-1-7-1-full-js-api-and-support-for-reversed-order-bits-lu-and-la/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] other end hangup
how can i disable other end hangup when i recive incomming call tfrom asterisk -- *Thanks&Regards,* *Jagadish Thoutam,* *IT Professional,* *+91-9963724776,* *jaganthou...@gmail.com.* ** -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Flash WAV Player
Hello, Does anyone know of a Flash or Java player that can play WAV files created by Asterisk? Thanks Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Security - Using Linksys PAP2T from outside with a dynamic IP is there anyway to block all other traffic but those of the PAP2T?
On Sat, 2 Oct 2010, Kyle Kienapfel wrote: > You're not going to be able to put a dns hostname in the iptables, but > you could have a script that runs at times and gets the ip address for > your dynamic hostname and allows that. Almost. You can put a host name in iptables, but it is resolved when loaded. You could restart iptables when your dynamic host name changes and it will be resolved correctly with the new IP address. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
