similar to: My Switch is being attacked using sip scanner tool (Service Abuse Attack)

Hello, I want to create a VPN between two Asterisk servers using OpenVPN. How to configure Asterisk and OpenVPN to do that. Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070809/ddaed76b/attachment.htm

Hello all, have anyone an idea about converting an audio file (.wav, .mp3, .au,...) to a ".gsm" audio file to use it as a voicemail file with Asterisk. Thanks. Abdelkader Mosbah -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070812/2ffc7135/attachment.htm

Hello, I have the following situation: A firewall is blocking all SIP and RTP traffic in the side of some of my clients. My clients cannot change settings of the firewall. I need to solve this problem and I need some help from you. I have this idea: implement a SIP user agent which does not use well known SIP ports (uses http port 80 for example) and use other ports that are not blocked

Hello All, MOS and R factor are the two QoS parameters used to estimate VoIP call quality. I have found that they are calculated from other metrics like jitter, latency, packet loss,...etc. But, haven't found any formula or arithmetic rule to calculate them. Do you have an idea about their formulas or an open source that calculates them. Is it possible to interpret them from wireshark.

Hello all, After installing Asterisk, i have installed the docs by "make progdocs". But i don't know where to locate this documentation. please Help. Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070810/ceb95948/attachment.htm

The failregex statement in my jail.conf file is: * failregex* = NOTICE.* .*: Registration from '.*' failed for '<HOST>' - Wrong password NOTICE.* .*: Registration from '.*' failed for '<HOST>' - No matching peer found NOTICE.* .*: Registration from '.*' failed for '<HOST>' - Username/auth name mismatch

What modules do you want on it? Yours, Michael Munger, dCAP 404-438-2128 michael at highpoweredhelp.com ________________________________ From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of MOSBAH ABDELKADER Sent: Saturday, August 04, 2007 3:16 PM To: asterisk-users at lists.digium.com Subject: Re: [asterisk-users] Connecting

hello, I have installed asterisk 1.2.18 in suse 10.2. After typing asterisk in the terminal command line (i don't think that asterisk runs when doing this) i type "asterisk -r" but the response" is "Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?)". how to solve this. thanks. -------------- next part -------------- An HTML attachment was

Hello, Is it possible to transfer ss7 signaling over an FXO interface. I need to setup an ss7 test system composed by two Asterisk based IP-PBX systems with anlog interfaces only (FXO and FXS). I want to know if it is possible to connect the two IP-PBX as following: - FXS interface in PBX1 -----------------> connected to -----------------> FXO interface in PBX2 =============>

Hello All, I have installed Asterisk 1.6 with openVPN in the same machine. I have set up a VPN connection between 2 SIP clients and Asterisk using x-lite. The 2 clients connects to Asterisk. SIP signaling goes ok over the vpn tunnel. When attempting to make a call between the clients, the siganling part of the call goes well. But, when the call is set up, some RTP packets are exchanged at

I noticed the following: Suppose that we have a server called "SMALLSERVER" working as a PDC for "SMALLDOMAIN**". When I enter "net getlocalsid" I get the following output: SID for domain SMALLSERVER is: S-1-5-21-xxxxxxxxx-xxxxxxxxx-xxxxxxxxx But when I enter "net getdomainsid" I get: SID for local machine SMALLSERVER is:

Hi, Try: ?vec1<-c(1,1,1,1,1,1,1,1,1) if(all(vec1==1)) "xxxxxxxxx" else? "yyyyyyyyyyy" #[1] "xxxxxxxxx" ?vec2<-c(rep(1,4),2) ?if(all(vec2==1)) "xxxxxxxxx" else? "yyyyyyyyyyy" #[1] "yyyyyyyyyyy" #or if(length(unique(vec1))==1) "xxxxxxxxx" else? "yyyyyyyyyyy" #[1] "xxxxxxxxx" ? if(length(unique(vec2))==1)

I'm seeing this: 1 UID FETCH 31734 (ENVELOPE) * 23 FETCH (UID 31734 ENVELOPE ("Fri, 2 Mar 2012 19:05:24 -0500 (EST)" "XXXXXX" (({22} XXXXX \"X-XX\" XXXXXX NIL "XXXXXXX" "XXXXXXXXX.XXX")) (({22} XXXXX \"X-XX\" XXXXXX NIL "XXXXXXX" "XXXXXXXXX.XXXXXX.XXX")) ((NIL NIL "XXXXXXX"

Hello all, I know this has been covered on list but can not find the answer I need, lots of references to no authority found, but none with an answer. I have two * servers, one behind firewall with nat the other on a dmz with nat. Both servers register with each other successfully. home is today's CVS-HEAD away is Asterisk 1.0.7 on away: Registered to '165.xxx.xxx.xxx', who sees

Hello, I am trying to connect an asterisk 1.6 to a trunking plate forme. With asterisk 1.4.x I added to sip.conf a line asking for registration in the form of: register = XXXXXXXXX at domain.com:Password:XXXXXXXXX at domain.com<assword%3AXXXXXXXXX at domain.com> @domain.com Unfortunately, as you can see, my usernames have to be of the form XXXXXX at domain.com which means that I had to

Is there a way to restrict connection to my asterisk server to users based on their IP addresses, and not just password. I have some hackers who connect to my server to make illegitimate solicitation calls to people. I had to shutdown the server for now until I find a solution. ANY HELP? Thanks. ond -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, What is the maximum number of simultaneous calls supported by asterisk. thks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090703/0794c554/attachment.htm

Hi, Thank you so much for your reply, I've a dedicated server in OVH, where I have done speed test for the server : *bkf at xxxxx ~> speedtest-cli Retrieving speedtest.net <http://speedtest.net> configuration...Retrieving speedtest.net <http://speedtest.net> server list...Testing from OVH SAS (x.x.x.x)...Selecting best server based on latency...Hosted by fdcservers.net

I *think* I may have encountered a bug, or a feature, in the idmap/winbind area. I have recently added rfc2307 attributes to my AD, and am in the process of switching over. This means that I still have (unintentionally) some files/directories/etc. around with old UIDs e.g. 3000007, rather than my rfc2307 specified UIDs. What I am seeing is that the SID2XID mapping is initially correct for a

I'm replacing a SoundPoint IP 600 with a SoundPoint IP 650. I attempted to simply reuse the existing config files for the old phone on the new phone, but the new phone would lock up on the 4th digit when attempted to dial out certain numbers. So, I downloaded the newest firmware and config templates from Polycom, and attempted to migrate the settings. Seems I'm missing something from