similar to: Call Quality Measuring

Hi Patrick, You are welcome to try our tools out for active and passive voice quality measurement tools. It's waveform analysis (like PESQ or POLQA) and VoIP metrics analysis (like G.107 E-model and other metrics). You can read more at http://www.sevana.biz or older site http://www.sevana.fi On Tue, Mar 31, 2015 at 1:16 PM, Patrick Beaumont < p.beaumont at hatsoffsoftware.co.uk>

Hi Patrick, try voipmon, there it's free and you can even track MOS. Markus Am 25.03.2015 um 14:21 schrieb Patrick Beaumont: > Hi everyone. > > We regularly get customers complaining about call quality issues. Most of > the time it turns out to be their own broadband. Very occasionally server > load. Does anyone have any advice or links to advice on measuring call >

Some SIP hardphones (Polycom) or softphones (Counterpath) embed a module that metter MOS. Regards 2015-03-25 14:21 GMT+01:00 Patrick Beaumont <p.beaumont at hatsoffsoftware.co.uk>: > Hi everyone. > > We regularly get customers complaining about call quality issues. Most of > the time it turns out to be their own broadband. Very occasionally server > load. Does anyone have

Thanks Richard. This is exactly the answer I was looking for. I'm now assuming that Asterisk 11 was using it's equivalent "bridge_simple" but I was getting confused because the only bridge module I saw in modules.conf was bridge_softmix. When I upgraded to Asterisk13 that would have been the only bridge getting loaded at first. Is it expected that if bridge_softmix handled a

Hi Everyone. I was referred here by malcolmd of the Asterisk forums. What follows is a copy of this question: http://forums.asterisk.org/viewtopic.php?f=1&t=92007? I've recently upgraded from Asterisk 11 to Asterisk 13. Most of it went smoothly thanks to the documentation detailing how to upgrade to 12 and then how to upgrade to 13. The only thing that didn't work correctly was

I am not sure if its just me, but i am able to get only local channel variables containing RTCP QOS values. The Version is 1.8.14. I want to store values of bridged channel in CDR. Phone is Cisco 7941 SIP and with sip show channelstats i see all the relevant information (jitter,packet loss) i want to get. It even calculates packet loss in %. But i am not able to store it to CDR. Asterisk 1.4

Hi, checkout new open source voipmonitor.org SIP packet sniffer.?I've developed it for my telco company and I've decided to share it. Testing and contributions are welcome! VoIPmonitor is open source live network packet sniffer which analyze SIP and RTP protocol. It can run as daemon or analyzes already captured pcap files. For each detected VoIP call voipmonitor calculates statistics

Hi there, I support a large number of enterprise users who contractually must connect to our support center via a 4G VOIP connection. I simply want to be able to auto detect all poor quality calls in realtme (as they are being made), play a message and drop the call - without user intervention. All decent call quality calls will be allowed through - to be handled by support staff. Its a

I am seeing lots of lost packets when running the command sip show channelstats at the CLI. There are issues across multiple Asterisk servers I am trying to diagnose but everything I read seems to point to this command being pretty unreliable. Can I trust the info this command shows? I am showing lots of lost packets in sip show channelstats but I can't see any packet loss when pinging the

I've seen something similar with Adtran SIP gateways. When a re-invite happens the Adtran gets all confused about call stats and marks the pre-reinvite leg of the call as losing large numbers of packets. BTW, IIRC reinvites happen when a codec changes or the channel switches to T.38. Also Adtran SIP gateways appear not to support OPTIONS packets when running in SIP proxy mode, which is

I would recommend capturing traffic outside your Asterisk server with Wireshark, then running the Telephony/Rtp/Analysize Streams option to determine if you have packet loss at that point in the network. On Mon, Jan 19, 2015 at 1:00 PM, Todd R. <tjrlist at live.com> wrote: > Thanks but no Adtran here. > > I do think these stats are indicating an issue, I just don't know how to

Hello friends, I am trying to deeply integrate asterisk cdr with voipmonitor cdr. I can access the caller Call ID (fbasename field in voipmonitor cdr) looking at the SIPCALLID variable in asterisk, but how can I access from within asterisk the Call ID of the second leg of the call (the one originating from asterisk to the destination peer)? is there a variable holding this value? Thank you

Hello, what could be causing the issue of poor sound quality ? Some calls, certainly not all of them, sound like if the caller is standing next to a very busy road with lots of cars passing. To be clear : the person calling is not standing next to a highway. But there seems to be a noise "on the line" of busy highroad that makes that the caller can not be understood. What can be

I want to capture all SIP messages. I have about 30 hosts in about 6 colos. My first thought was dumpcap, but the output file name format bugs me. What do you use for long term SIP capture? -- Thanks in advance, ------------------------------------------------------------------------- Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST

The bar is getting raised yet again http://www.voipmonitor.net/2007/06/05/New+Features+And+Services+For+Pack et8+Virtual+Office.aspx I personally use Snapanumber $30 or there abouts (after trialing a few other TAPI solutions and finding them sub-par) and think it's a great product but interesting to see how more people are expecting desktop/phone integration applications. Does anyone

examples of "interesting" information like ICE result and howto make "minimal" configuration of pjproject.conf i.e. for  debugging app_queue.so core set debug 5 app_queue.so for debugging RTP core set debug 10 rtp_engine core set debug 10 res_rtp_asterisk rtp set debug on logger.conf rtp => debug,verbose(5) so i mean in

Ok. I just got a TDM10B and it is in with my X100P. So as it says in the provided instructions, I used the command modprobe tor2 I get an error message saying that there is no such device. My zaptel.conf looks like this: fxsks=1 fxoks=2 So I load the X100P first. (modprobe wcfxo) Then I load the TDM10B (modprobe tor2) Then I'm told that the device doesn't exist. Please help

So I have an X100P, and a TDM10B both working (at least I think they are). The drivers have been loaded and ztcfg -vv shows no errors in the configuration of two channels. When I run asterisk -vvvc and pick up my phone (plugged into TDM10B), I don't gear a dialtone. in phone.conf, I have [interfaces] mode=dialtone format=slinear ... Shouldn't that produce a dialtone when I pick up the

Here's the problem: I start asterisk, and it takes up around 3-4% of my CPU resources. However, this number continues to climb over the hours until it is close to 100%. Usually it takes around a day to climb up to approximately 95 or 96% Has anybody experienced the following problem before?

My setup: X100P and Quicknet PhoneJack. I can't seem to properly set up a Zap channel for my X100P. Here are some of my configurations: [zaptel.conf] fxsks=1 #X100P fxoks=2 #Quicknet PhoneJack defaultzone=us loadzone=us [zapata.conf] [channels] context=local signalling=fxs_ks channel->1 ;X100P [extensions.conf] ... [local] exten=>_NXXNXXXXXX,1,Dial,Zap/1 ;I'm pretty sure the