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...s at lists.digium.com> on 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 upgrade...

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

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

...have been using. For additional information: gc() gives Garbage collection ... 123580 cons cells free (61%) 1738k bytes of heap free (89%) I am very grateful for your help and advice. Do you have any more suggestions? Do you think there is a simple patch for paste? All the best, Mark Mark A. Beaumont Institute of Zoology Zoological Society of London Regent's Park London NW1 4RY UK Tel: 0171 449 6617 Fax: 0171 586 2870 m.beaumont at ucl.ac.uk -.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.- r-help mailing list -- Read http://www.ci.tuwien.ac.at/~hornik/R/R-F...

...ve installed around 3 or 4 previous versions on the same system and never had a problem (but, then the Makefile was very different). The first make to compile everything went OK as usual. I have a standard installation of perl 5 (accepting all defaults in configure). Any ideas? Mark -- Mark A. Beaumont Conservation Genetics Group Institute of Zoology Regent's Park London NW1 4RY Tel: 0171 449 6617 Fax: 0171 586 2870 Email: m.beaumont at ucl.ac.uk -.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.- r-help mailing list -- Read http://www.ci.tuwien.ac.at/~hornik/R/R...

...ve installed around 3 or 4 previous versions on the same system and never had a problem (but, then the Makefile was very different). The first make to compile everything went OK as usual. I have a standard installation of perl 5 (accepting all defaults in configure). Any ideas? Mark -- Mark A. Beaumont Conservation Genetics Group Institute of Zoology Regent's Park London NW1 4RY Tel: 0171 449 6617 Fax: 0171 586 2870 Email: m.beaumont at ucl.ac.uk -.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.- r-help mailing list -- Read http://www.ci.tuwien.ac.at/~hornik/R/R...

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

...ols 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> 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....

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?

...2]: loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/app_realtime.so: undefined symbol: ast_load_realtime Jan 7 19:17:22 WARNING[1752]: loader.c:440 load_modules: Loading module app_realtime.so failed! foxy*CLI> Disconnected from Asterisk server root@foxy:~ > Best Regards Steve Beaumont

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

Here is the situation that I would like to create: Call comes in Receptionist sees that the caller ID is Jenny <8675309> Receptionist picks up phone and transfers call to Batman Batman looks at his phone and sees that the caller ID is Jenny <8675309> I can't seem to figure out how to forward the caller ID. Is this possible with Asterisk?

...this I got group/ownership errors with tar and the make failed. So then (following the previous patterns) I installed GNU tar. "make install" then worked fine, But I still get the same problems when trying to run R. I think I'm almost there....Any ideas ? Best wishes, Mark Mark A. Beaumont Institute of Zoology Zoological Society of London Regent's Park London NW1 4RY UK Tel: 0171 449 6617 Fax: 0171 586 2870 m.beaumont at ucl.ac.uk -.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.- r-help mailing list -- Read http://www.ci.tuwien.ac.at/~hornik/R/R-F...

I need help creating a channel for my phone device (Quicknet PhoneJack). I have installed and loaded the driver and phone devices listen in /dev (phone0 - phone15). [phone.conf] mode=dialtone format=slinear device => /dev/phone0 fxoks=2 ;Quicknet PhoneJack [extensions.conf] ... exten=>_NXXNXXXXXX,1,Dial,Phone/phone0 ... When I try to make a call, I get the following output: Executing

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

Does anybody know a place where I can download a working PWLIB and H323? The files at www3.openh323.org are not in gzip format (according to the error message I receive) gunzip: pwlib_1.4.11.tar.gz: not in gzip format I have also tried using the tar -xzvf command, and that gives the following error: gzip: stdin: not in gzip format tar: Child returned status 1 tar: Error exit delayed from

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...2 This is with gnu make. I get the same thing with the DEC make. It looks as though it may be something to do with src/library/LIBSETUP, which is certainly rather different from the version in 0.61.1. (R-0.61.1 made OK, but only with gnu make not DEC make. ) Any ideas? Best wishes, Mark Mark A. Beaumont Institute of Zoology Zoological Society of London Regent's Park London NW1 4RY UK Tel: 0171 449 6617 Fax: 0171 586 2870 m.beaumont at ucl.ac.uk -.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.- r-help mailing list -- Read http://www.ci.tuwien.ac.at/~hornik/R/R-F...

I was playing with the agent application to see if I could get it to work. Everything works fine, except that Asterisk plays musiconhold while an agent is logged in and is not taking a call. Is there a way to disable the music in this situation? Imagine working tech support where you had to listen to hold music when you weren't taking a call. Now think of your company's choice of hold

I used to have email notification working with my voicemail services but it stopped working when I installed the new version of asterisk. I have not changed my voicemail.conf file, so I'm out of ideas. Does asterisk use Sendmail to send messages, or does it have its own method for sending email?