similar to: RE: Coaching in asterisk

BTW. We only use Asterisk for a few functions. Everything else is done on an extenal application controlling Asterisk through AMI. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Wai Wu Sent: Friday, March 09, 2007 12:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE:

NVWhisper. Justin ------------------------------ Date: Thu, 08 Mar 2007 16:25:28 -0500 From: Wai Wu <wkwu@calltrol.com> Subject: [asterisk-users] Coaching in asterisk Is there a way to setup a conference where party A can coach another Party B, at the same time, all other parties cannot hear party A? In order words, partis A and B can hear every one, and party A can only be heard by

Anyone tried the digium TC400B transcoding card? What are your opinions? Thnx

It sounds like Naren and company has their own CRM application. They need a predictive dialer that allows third party app integration. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of Matt Florell Sent: Monday, March 13, 2006 8:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re:

Hi all, Just ran into some issue with the originate AMI command. It seems that there is a limit of around 120 calls I can place with the originate command simutanously. By that I mean sending Asterisk a lot of originate command very fast. Anyone know if there is a limitation? Thnx. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I need the zap channels going, but got the following error. What do I need to change in my configuration? Thnx. chan_zap.c: In function `zap_send_keypad_facility_exec': chan_zap.c:2309: warning: implicit declaration of function `pri_keypad_facility' chan_zap.c: In function `pri_dchannel': chan_zap.c:9292: structure has no member named `call' make[1]: *** [chan_zap.o]

Hi All, I'm conducting a meta-analysis and have taken a data.frame with multiple rows per study (for each effect size) and performed a weighted average of effect size for each study. This results in a reduced # of rows. I am particularly interested in simply reducing the additional variables in the data.frame to the first row of the corresponding id variable. For example:

Dear R Community, I am attempting to transpose a dataset from rows to columns but am stuck. I have tried using reshape() with little luck, possibly due to the categorical nature of the data. For example: id<-c(1,2,2,3,3,3) author<-c("j","k","k","l","l","l")

Except that mixmonitor still has a bug in it. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Kevin P. Fleming Sent: Wednesday, April 12, 2006 11:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] call center running Asterisk -sound quality-critical! Matt Roth

Hi all, Any one know how to increase the Linux limit? I am hiting a wall on 200 calls playing files at the same time. From Asterisk console, I am getting messages like Sip_request_call: Unable to build sip pvt data for "asterisk1/700" Too many open files Is this a limit of my Linux box? I only have 512MB of ram. Will increase it to 2G help or I have to change some configuration in

I just check the source code, Monitor uses ast_writestream and it eventurally goes down to au_write, g723_write, etc. They don't commit to the disk. So, in effect, if you have a lot of ram, the audio should stay in ram until it gets swap out or the file is closed. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On

Hi everyone, I am running into wall today with simultaneous call limits. I have two Asterisk machines (fast 3GHz C2D with 2GB of ram). I tried to create a lot of sip calls from one machine to the other by issuing AMI Originate commands to one machine. The machine that makes calls plays a message (demo-intruct) upon the other machine answer. The machine receives the calls just waits for 40 seconds

Hi all, Is there a way to connect two live calls through the manager api without directing them to a meeting room? Currently, I can connect them by sending them to a meeting room. However, I don't know what the overhead is, and I kind of think that if I can connect them or link them up, the overhead would be minimum. -------------- next part -------------- An HTML attachment was scrubbed...

I did not install soxmix in my linux box. If you having issues with mixmonitor, you can put both legs of the call into a conference and record the conference -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Matt Roth Sent: Thursday, April 13, 2006 1:20 PM To: Asterisk Users Mailing List - Non-Commercial

Hi, I was thinking about bidirectional sync of sysvol and i've a question: ?What about DRBD?. You can create a disk partition in every node, create a DRBD cluster and then mount that partition on sysvol folder. The sincronization is bidirectional and in real time. For now i've not tested this option, but i've plans to start some tests. What is your opinion about this? Greetings!!

Dear Daniel, Klaus I've try that before But because of how samba work on the files. The Advise is No Without CTDB, you will just shoot yourself on the foot... On Thu, Jun 25, 2015 at 7:39 PM, Zerwes, Klaus <zerwes at rosalux.de> wrote: > Just some notes: > For master <-> master setup (bi-directional sync) you need AFAIK a cluster > filesystem. > I have no idea

Dear Belle, That produce the similar situation. Thus I'm trying osync And the result are much present according to my test case And configuration are much streamline. However, I'm not sure how it can work with 2 DC or more... Thank You On Mon, Jun 22, 2015 at 9:05 PM, L.P.H. van Belle <belle at bazuin.nl> wrote: > Hai Min Wai Chan, > > I have tested it as shown in

My TE411p does not seem to like to have two PRIs from different telcos (span 1 and span 2). I can get one working, but not the other. However, if I use vpmsupport=0 when loading the wct4xxp module, they both work. But here is the problem, vpmsupport=0 disables the on board echo cancellation. Any ideas? BTW, here is zaptel.conf span=1,1,0,esf,b8zs span=2,2,0,esf,b8zs bchan=1-23 dchan=24

Does Unision support ACL? If it would than this part guide would work. https://wiki.samba.org/index.php/SysVol_Bidirectional_Replication But it didn't.... On Mon, Jun 15, 2015 at 7:14 PM, Bob of Donelson Trophy < bob at donelsontrophy.net> wrote: > > > For bidirectional sync take a look at Unison (file synchronizer.) > > --- > > ------------------------- >

2015-06-25 14:44 GMT+02:00 Daniel Carrasco Mar?n <danielmadrid19 at gmail.com>: > > > 2015-06-25 14:12 GMT+02:00 Min Wai Chan <dcmwai at gmail.com>: > >> Dear Daniel, Klaus >> >> I've try that before >> But because of how samba work on the files. >> >> The Advise is No >> Without CTDB, you will just shoot yourself on the