similar to: MOS Scores and LCR

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, 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

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

Asterisk handling My Skype Calls This is for me, once more, Asterisk as the Future of Telephony. Today I've integrated my Skype Account as SIP extension in my * Box. This has been possible using "Uplink Skype to SIP Adapter", available for free at http://www.nch.com.au/skypetosip/index.html . Main features that any one can easily integrate into Asterisk: - Route skype incoming

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

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

Is there a list somewhere or a way to find the following: 1- All non US 48 area codes which can be dialed as 1+10 2- All strange area codes which are used for premium services such as 900-XXX-XXXX 3- Anything else that should be restricted if one was to restrict all calls to US 48 only I have found many list but it's tough looking at the entire list of area codes and pulling out each of them

I saw several threads requesting an Asterisk forum to complement the email list. i.e. http://lists.digium.com/pipermail/asterisk-dev/2004-February/003103.html I recently created an Asterisk forum within TMC's popular VoIP forums for everyone to use. http://voip-forum.tmcnet.com/voip-forum/forum/forum_topics.asp?FID=15

List, Does anyone know how to add the dst Country to the CDR's via Macro (preferably). For example, I will add a column in the cdr DB table and when someone dials 01158212XXX. I want the CDR's to show Caracas as the destination in this new column. I have all of the International destinations in my extensions.conf like the example below: [macro-dialout-intl] exten =>

How to set-up LCR ? a. which companies can be used with LCR? b. how to set-up & maintain LCR? c. multiple connection to one gateway? Example: +886223456789 could be reachable via a. ENUM free b. Dundi free c. Voipstunt free d. Voipbuster free e. Nufone $ f. Voipstunt $ g. others with 4 concurrent connections $$ h. others with 3 concurrent connections $$ I am looking

Hi, After a few extra days of hard work, debugging, and many coffees, I am pround to announce that Asterisk::LCR has been released on CPAN. Asterisk::LCR is an open-source, Perl-based collection of tools to help you manage efficiently multiple VoIP providers with your Asterisk installation. It is capable of importing providers rates from multiple providers, comparing these rates, and

Hi all, I am planning to implement LCR routing on my already running asterisk server. Uptill now i have found out that asterisk has no support for lcr, i have to do something about it myself, for example using the AGI. Im looking for ideas here. Whats the best way to start implementing lcr in asterisk. Should i use agi and start implementing my own lcr script or is there any plugin available which

Hi List I need a quality US 800 DID over IAX for my Asterisk server, preferably one that doesn't cost the earth. Any suggestions please? Thanks -- Chris Blunt Entropy IT Ltd -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061220/4919f3cb/attachment.htm

Hello, I short question: I want to connect Asterisk to OpenBSC with mISDN, mISDNuser and LCR. Do I need chan_lcr? I have: Asterisk 1.8 mISDN .v2 integrated in Kernel 3.0.22 mISDNuser lcr 1.7 HFC-E1 Evaluation board from cologne chip I tried to configure Asterisk with <./configure --prefix=/usr/src/lcr --with-gsm-bs> and it runs without errors, But make and make install didn't run

I have got Areskicc installed with AMP and I can't stress out how good and excellent this software is. I must say the author desire every credit for this software and I would like to say Thank you to Arezqui who wrote the software. Now there is one last thing before my whole asterisk calling card become prefect. And I am wondering whether anybody may know the answer of this. I

It will be interesting to see how many standards get broken, and how many proprietary hooks get thrown into the pot. The bean counters smell some money, and their OS franchise is waning: http://www.nytimes.com/2006/06/26/technology/26soft.html -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean.

Hi, This is what I want to do: 1. Asterisk to answer calls via DID's, currently using SIPGATE 2. Provide a menu, and allow users to dial out. 3. According to the country and area they dial, the call should connect via one of up 4 carriers depending on cost. 4. If the carrier is busy it should go to the next one in line and so forth. I have tried to set this up, but it never answers the

Hello asterisk-users. I have the following dial-plan: [test] exten => 482,1,Dial(OH323/106@192.168.2.73,10) exten => 482,n,Dial(OH323/102@192.168.2.73,10) exten => 482,n,Dial(OH323/103@192.168.2.73,10) exten => 482,n,Dial(OH323/104@192.168.2.73,10) exten => 482,n,Dial(OH323/105@192.168.2.73,10) exten => 482,n,Dial(OH323/106@192.168.2.73,10) When I call exten =>

Dear All, I have two questions. 1. How can I distribute equally the outgoing calls between channels in a group? I'd like to send outgoing calls on 2 ISDN GSM adapter. I need to use the GSM subscriptions approximately equally. 2. When all of the channels are busy in the first outgoing channel group. How can I send out the call on another group? N+101 is for this problem? Regards, Repas