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...n behalf of Richard Mudgett <rmudgett at digium.com> Sent: 09 December 2014 20:49 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Bridge configuration in Asterisk 13 [Spam score:8%] On Tue, Dec 9, 2014 at 1:35 PM, Patrick Beaumont <p.beaumont at hatsoffsoftware.co.uk<mailto:p.beaumont at hatsoffsoftware.co.uk>> wrote: 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...

...ve 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> wrote: > Thanks for the suggestions guys. I?ll try to have a play with Voipmonitor > in the near future. > > So can I assume from the lack of discussion nobody is using the ?sip show > channelstats? stuff? > > Regards, > Patrick. > > On 31/03/2015 08:23, &q...

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

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 quality? I?ve been playing around with ?sip show channelstats? but can?t other than measuring the packet loss I don?t really know what I?m supposed to be looking for

On Tue, Dec 9, 2014 at 2:58 PM, Patrick Beaumont < p.beaumont at hatsoffsoftware.co.uk> wrote: > 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_s...

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 any advice or links to advice on measuring call > quality? > > I?ve b...

On Tue, Dec 9, 2014 at 1:35 PM, Patrick Beaumont < p.beaumont at hatsoffsoftware.co.uk> wrote: > 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. >...

Hi Everyone. I have a customer looking to deploy about 20 Grandstream GXP2140 phones. Normally they would deploy Yealink brand phones but they want a phone with gigabit pass through and the Yealinks with gigabit are too expensive for their budget. Does anyone on the list have experience with the GXP2140? Is it a reliable phone? Does anyone have recommendations for other phones with gigabit

Have you enabled DTMF logging and seen the DTMF codes being recognised by Asterisk? I had a bunch of soft phones that I had to change to using ?sip info? for the DTMF signalling as the RFC signalling was not always being recognised. This would cause transfers to appear as if the user had not dialled any digits. On 20/12/2014 20:52, "sean darcy" <seandarcy2 at gmail.com> wrote: