similar to: Bridge configuration in Asterisk 13

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

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. > > Most of it

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_softmix. When I

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

Hi, I am trying to use ConfBridge application, but it throws "Failed to find a bridge technology to satisfy capabilities 0x4 (ulaw)" error. Please see console output below. -- Executing [501 at services:9] ConfBridge("SIP/OpenSER-00000005", "1001") in new stack [May 19 13:36:05] DEBUG[7452]: app_confbridge.c:404 join_conference_bridge: Trying to find conference

Hi All, I am working on Asterisk 12.6.0 with ConfBridge module, when there are multiple user like admin and normal participant running with conference. When I try to kicked 2 user (Normal User), it play file "conf-kicked" and again join conference My scenario in confbridge like. 1] Admin User (e.g. SIP/8484-00000000) 2] Normal User (e.g. SIP/8484-00000001) 3] Admin User (e.g.

Hello, I am struggling with a problem which I thought would be an easy one : bridging several channels together in a *smart* bridge. I emphasize *smart* : I want my bridge to be a native_rtp one when only two channels are involved, and switch to softmix technology when a third channel comes in. I thought I could use ConfBridge for that, but it creates a bridge that is not smart (it is of

After an upgrade to Asterisk 12, I'm "collecting" channels. When I enter and then exit a conference room, I see: -- <CBAnn/207-0000067f;1> Playing 'confbridge-leave.slin' (language 'en') -- Channel CBAnn/207-0000067f;2 joined 'softmix' base-bridge <5edb1920-3774-4ba3-8c4d-23e8fd04519c> -- Channel CBAnn/207-0000067f;2 left

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>

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

The Asterisk Development Team would like to announce the release of Asterisk 16.0.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 16.0.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release:

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

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?

The problem is the following # make docs Make: Don't know how to make ../src/library/*/man/*.Rd. Stop. *** Exit 1 (ignored) # I am using a DEC alpha 200/4/233 with Digital Unix 3.2D My previous version of R was R-0.50-a1. I have 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

The problem is the following # make docs Make: Don't know how to make ../src/library/*/man/*.Rd. Stop. *** Exit 1 (ignored) # I am using a DEC alpha 200/4/233 with Digital Unix 3.2D My previous version of R was R-0.50-a1. I have 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

Douglas Bates wrote: >From 0.62 onward you should not have to create a symbolic link in >/usr/local/bin. It should be that you can run > cd $RSOURCE > ./configure --prefix=/usr/local > make install >and you will end up with the R script installed in /usr/local/bin and >all the files needed to run R in /usr/local/lib/R. > >Can you tell what the prefix is set to after

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

The Asterisk Development Team would like to announce the release of Asterisk 13.30.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.30.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release: