search for: talker

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 Use...

...10P 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 saying. When talker A stops talking, he/she can then hear what talker B says. This issue occurs across all the different phones we have set up. I have played with the OSLEC settings in the thoughts that the echo cancellation was being a bit ambitious, to no avail...

...t unsure as to when one uses a nested resource and when one uses a normal resource. If you have a belongs_to, has_one/many relationship between models is that automatically an indication 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...

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.

....000000000 -0400 *************** *** 2901,2905 **** to write out all the samples. */ ! if (user->talking && !(confflags & CONFFLAG_OPTIMIZETALKER)) { careful_write(fd, f->data.ptr, f->datalen, 0); } --- 2901,2905 ---- to write out all the samples....

...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 traf...

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 busydetect=no, are there any other helpful bits to specify? Sincerely, Brent A. Torrenga brent.torrenga@torrenga.com Torrenga Engineering, Inc. 907 Ridge Road Munster, Indiana 46...

...eb UI Options I m working on : * call out and join new participants to the conference room * ... At the end of the email, you will find the zatpel conference mode, I just implement the two masks (only listen and listen-talk): - ZT_CONF_CONFMON | ZT_CONF_LISTENER - ZT_CONF_CONF | ZT_CONF_TALKER | ZT_CONF_LISTENER Anyone needs more ?!? Well, I wanted to ask to the list if some of you are interrested to it (hope to) and if perhaps if you have some ideas concerning options that we can add or whatever to make this application better. And if anyone need special custom/features, feel free a...

...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 separatel...

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 more, displaying all DTMF received,...

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

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 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'...

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 the confbridge? How do I tell whats up? Thanks Jerry -------------- next part ----...

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

...others say ... yes, but .... yes, but... no, because.... You need this feedback. People are iritated if the call someone how have a phone that mutes in nobody is speaking. Youz have the feeling that you speak with a wall. That's it. > 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 latency. True, but unless you change The internet, there are not much posibilities....

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