similar to: Calculating R Factor and MOS metrics for VoIP

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, 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

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

An attacker is scanning my Asterisk Switch to gain illegitimate access to VoIP call functionality. Using a sip scanning tool, *it* sends REGISTERs with random identities. And when it discovers one identity subscribed in my switch, it tries to authenticate with random passwords using this user name. For the moment, I have replaced this account. And also blocked the IP it has used but each time

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

Hello all, I'm tring to retrieve a formula to calculate a MOS value from Asterisk RTCP stats... Have you got any idea how to do it? Thanks I'm reading all G.107 ITU docs to retrieve something... I'm saving the SIP RTCP stats with: [macro-hangupcall] exten => s,1,Set(CDR(userfield)=${CHANNEL(rtpqos|audio|all)}) exten => s,n,ResetCDR(vw) exten => s,n,NoCDR() So I retrieve

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

Is there any tool that can do LCR for Asterisk but also take into account MOS scores? Is it possible to automatically generate MOS scores on random "calls" so as to keep an updated database on a per provider, per destination, per time-of-day score? Hopefully, with that information we can create a better LCR module or script? Thanks, Daniel

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, I just signed up to this mailing-list (note: my first mailing list at all), because I'm having some problems related to speex. Let me just introduce you to what I'm doing. I am writing a short (really short) paper about VoIP techniques, especially audio codecs for speech. I pointed out basic technologies behind audio codecs; vector quantization, lpc, long-term prediction and some

Hi, I've upgraded the driver provided by the manufacturer of the C-MOS [1] UPS, which is an argentinian company. The file I'm attaching works with nut 2.0.4, I have one C-MOS UPS and I've succesfully tested it. It would be nice to have it included in the nut codebase. [1]: http://www.c-mos.com.ar/ -- Regards, Maxy -------------- next part -------------- A non-text attachment was

Hi, I've got a problem with booting zfs-on-root FreeBSD 9.2-PRERELEASE. I'm getting: ZFS: i/o error - all block copies unavailable ZFS: can't read MOS of pool klawisz gptzfsboot: failed to mount default pool klawisz Machine is VM running under KVM on Proxmox 2.3-13. VM has 8 GB of RAM, 400 GB of local storage with SCSI Controller type: Default (lsi). I'm not sure what I did to

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

Hi, Although this is a repost from Asterisk biz, we would like to ask if somebody may help us to develop a native Asterisk module using AQuA technology for voice quality monitoring using the same web service AQuA Meter is using. Thanks, Sevana Finland/Estonia ---------- Forwarded message ---------- From: Sevana Oy <sales at sevana.fi> Date: Mon, Jun 17, 2013 at 7:30 PM Subject: AQuA Meter

I have been in the process of trying to troubleshoot a phone system that is doing IAX trunking to a provider. The average RTA is 75ms with spikes from time to time and jitter from time to time as well. My question is this; How much can one trust this types of samples when going over the internet? I mean who knows who is doing what kind of ICMP rate limiting or dropping ICMP all together? What is a

Dear all, I am using MOStest in vegan to examine the hump-shaped of a quadratic regression model. In the manual of vegan, it mentions that the value of MOStest include isHump. My result showed the quadratic regression is hump-shaped according to the p-value (nearly 0). However, the value isHump did not show up. Please kindly help where to find isHump. Further, I do not understand the result of