search for: talkers

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

...of a nested resource or can these resources still be represented in the normal resource way? I have a resource (talker) that belongs_to a number of other models (network, data_date, talker_type etc) in a many to one relationship and I''d like to be able to have something like: /networks/1/talkers/10 /data_dates/1/talkers/10 /talker_types/1/talkers/10 but also /talkers/10 Is it possible to do this? or do I need to rename each of the nested parts of the url to something like network_talkers etc. so that it would be: /networks/1/network_talkers/10 ? Thanks, Toby --~--~---------~--~----...

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 decoding NFS who is doing what.

Is this patch correct? The "&&" doesn't make logical sense to me. I think it should be "||" and making this change fixes the problem I have with SIP phones in MeetMe conferences. If it's correct, is there someplace more formal that I should submit it to? *** app_meetme.c.old 2009-10-11 17:56:44.000000000 -0400 --- app_meetme.c 2009-10-17

...acket loss on long-haul IAX2 trunks (i.e.: satellite) - load testing G.729, ILBC, Speex, and other complex codecs as a relative comparative load on a dual 3.0ghz Xeon machine (quantitative testing, not seat-of-pants testing) - maximal traffic density on 802.11[a,b,g] links with multiple IAX2 talkers (trunk mode) - maximal traffic density on 802.11[a,b,g] links with multiple IAX2 talkers (normal mode) - maximal traffic density on 802.11[a,b,g] links with multiple IAX2 talkers (routed) - maximal traffic density on 802.11[a,b,g] links with multiple SIP talkers (normal) - maximal traff...

Hi all, After loading 1.6.0.1, I notice that I always have the "VOX" effect on Meetme conferences whether I have the "o" option set in the dial plan or not. Is anyone else seeing this? Although I'm now running 1.6.0.1, I'm also seeing this on a system still running 1.6.0beta9. Thanks. -- Bill in Denver

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

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

...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 seconds to 27 seconds, then channel 1 again goes from 27 seconds to ... you understand. Then you have two channels, one channel is always silent/mute, the other contains the current talker. Then you can AGC them separately...

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 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

Sorry, got the subject wrong in the last mail. My e-mail client is playing games with me...

...o 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 making it louder by factor of 2.0would be necessary). My question: Is there a possibility in asterisk-1.4 to double/quadrouple the loudness of a channel's input/output using chan_sip? All clients come in via...

Hi Can someone explain me what is the purpose for MeetMe Option 'o'.. If I defined 'o' with MeetMe option or If not defined with MeetMe option... What is the difference between these two if defined or not defined MeetMe 'o' option... -- Regards, Chandrakant Solanki -------------- next part -------------- An HTML attachment was scrubbed... URL:

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 all, <p>once again i came up with my conferencing stuff. On a conference with more then two people it's a waste of bandwidth, that every entity send it's data to every entity. Since there is only one audio line, the audio must be merged on the server. Here are my questions: - How many audio chanels can a server process (let's say a 3GHz machine) in this way: decode all

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

Hi List, When a user is on a call, sometimes they hear digits dialing as if the other end is randomly pressing the keypad with their face...but they aren't. It has happened while I've been on calls also, very odd and annoying. Has anyone come across this on Asterisk before? TIA, Skyler

Hi all; I have been searching for an answer to a question that a customer asked me and I have only found a few older answers. So, wanting to find out if anyone has any experience with this issue and can help provide me with some advice. I have a customer which is strongly interested in using Asterisk as a PBX. One of the core requirements, however, is that the system MUST be able to