similar to: Nested Resources vs. Normal Resources

[moved to asterisk-users, as this is not a development question] At 1:40 PM -0400 on 4/24/04, Sudhir Kumar wrote: >I would like to hear from any of you who has done any kind of >benchmarking on a robust hardware that can handle large call volume, >preferably with G.729 codec involved. > >We are in the process of putting together a system that should have a >quad E1 card, G.729

We started seeing odd behaviour with clients somehow hammering our ZFS-based NFS server. Nothing is obvious from mpstat/iostat/etc. I''ve seen mention before of NFSv3 client dtrace scripts, and I was wondering if there ever was one for the server end, displaying top talkers, writes/reads, or locations of such to nail down abusive clients short of using snoop/tcpdump to nail down via

2011.08.23. 15:38 keltez?ssel, Yanick Bourbeau ?rta: > Since I don't have access to different channels as I record a phone call > using a man in the middle approach, there is something else I can use > to equalize the sound ? What I would do then probably is try to manually separate the two channels/talkers; say channel 1 goes from 0 seconds to 13 seconds, channel 2 goes from 13

Hello All This or similar topics have already been mentioned but without any solution yet. I have built a oneway conference system for a client using one caller's input and broadcast it to all the other participants using app_meetme. E.g. one talker multiple listeners. Unfortunately some of the talkers (I have got multiple rooms) are not loud enough (e.g. use just half the amplitude, so

Did anybody get this error message before: chan_zap.c:5044 do_monitor: select return -1: Bad file descriptor When it's happening, Asterisk gets freezed and talkers can not hear each other. This message appears like in a loop at the server's screen. thank you Oz _________________________________________________________________ MSN Messenger: instale grátis e converse com seus amigos.

Dear Members of Ruby n Rails Group! I''ve get involved to develop and manage The First Ukrainian Ruby on Rails Conference 2008 (Kyiv on Rails 2008 - Projectname). I''ve created a group - "Kyi''v Kolijamy" (Kyiv on Rails Conference) located at http://groups.google.com/group/Kyiv-on-Rails. Ukraine is located in Eastern Europe, so, if the marketing strategy woud say,

I have a recorded conversation from an analog trunk. As usual one side is stronger that the other one. In my case, the gap between signal levels are even bigger. How does speex AGC preprocessor will perform on this type of audio recording? Maybe I am wrong and AGC is not really what I need to equalize the two persons in my phone conversation? As I Understand, AGC will perform better if each

Hi there, I am trying to get the sum of rain per day. That is what the data-set looks like: Timestamp Rain_mm_Tot 2017-05-29 23:40:00 4.7999980 2017-05-29 23:50:00 1.2000000 2017-05-30 00:10:00 2.5800000 2017-05-30 00:20:00 1.2009600 2017-05-30 00:30:00 1.2000006 2017-05-30 00:40:00 2.5002480 First I tried to define the

Hi there, I would like to enable the desktop user to create shared folders, for example using protocols like Samba, nfs and sftp. For the user to make this possible, the user needs (on a Linux machine configured with the defaults of most current Linux distributions) root privileges. However. In a desktop environment, in my humble opinion the possibility for a normal user to configure a simple

Hello, all. I've been playing with MeetMe and talker optimization seemed like a great idea. I activated it as follows: exten => 201,1,MeetMe(100201,cTo) However, although I can see who is the talker on the CLI pbx01*CLI> meetme list 100201 User #: 01 1001 Denise Dion-Sullivan Channel: SIP/1001-1e1db7c8 (not talking) 00:00:33 User #: 02 1000 John A. Sullivan III

Hello, I am working on a trixbox based system with a TDM410P connected to 3 phone lines from the CO. The asterisk box is on a full duplex 100Mb LAN with some polycom and Aastra SIP phones. In general everything works. the problem I am trying to solve is that if both parties to a call speak at the same time one of the voices gets cut out such that the talker A cannot hear what talker B is

Dear Community, I recently started thinking about how to make a project like CentOS more transparent and open (especially for new contributors). The http://wiki.centos.org/Team page (which Dag created about a year ago) lists about 20 (more or less active) members, divided into core and community contributors. I personally do not like that kind of distinction. Of course there should be something

Well, I thought and hoped my issue of random hangups on our TDM400P were related to busydetect=yes in zapata.conf. The behavior of a call being hungup has not changed, however, since setting the busydetect option to 'no'. Again, the only affected user is my loud talker... What are some causes/solutions to seemingly random call disconnects on Zap channels that people have seen? I have

what anti-spam for you used ? dspam?spammassian? amavisd-new ? what is best ?

Hello Asteriskos, Screenshot: http://www.areski.net/asterisk-meetme/about.php The goals of this application is to control your audience/users in the conference room. That will allow you to have a visual presentation and to control the conferences over the net. A lot of changes has be made to app_meetme to keep some conferences informations into a DB and to check through if some properties has

Hi, What''s the best way to add support for the activeresource find method in the controllers in my rails app? As I understand it when you invoke something like: Person.find(:all, :params => {:name = ''toby''}) this generates a request of: GET /people.xml?name=toby Does this mean that in my index method in PeopleController in my rails app I have params[name] set to

Hi all, I have a Diva Server V-BRI-2 card, which support, as written in reference guide: Extended tone processing (human talker detection, generation and detection of country-specific tones) I would like to detect human speech and fax tone with asterisk. I think that the diva card transmit a DTMF code when detecting voice, but chan_capi doesn't receive this DTMF code. I verbose it

I have used every asterisk 11.8.X version. Have not had an issue till 11.8.0 and 11.8.1 When I use ConfBridge I am attempting to put all participants in MUTE mode and just one talker... However, since 11.8.0 I am hearing feedback in the announcement like the channel is not really muted. I dropped back to 11.7.0 and I hear no feedback. Has something changed - or - am I not correctly setting up

Hi Allen, > I tend to disagree. It normal human conversation it wouldn't make much > sense to have 2 people talking over each other at the same time. One of the problem is, that if the server doesn't distribute the stuff, then one entity must send the stream to every other entity. That could work fine with fast connections, but doesn't work with a modem connection. My

I tend to disagree. It normal human conversation it wouldn't make much sense to have 2 people talking over each other at the same time. Thus, it most scenarios you would have only one talker anyway. Additionally, encode->decode/mix/encode->decode isn't a very efficient CPU process for a server, it's complicated to keep timing correct and it has a negative impact on total