similar to: Local voice feedback

I'm not sure where to check next, so I'm reaching out to those that know this stuff better than I. I've got Asterisk up and running, but I've still got an occasional audio issue. Once in a while (maybe 1 out of every 20-30 calls), the audio becomes heavily distorted, but only on the local side. The party on the other end says the audio is fine. We can hear them, although

I have Asterisk running with a combination of SIP and H323 clients. I am using the OH323 module instead of the H323 one. When the SIP clients ring each other, they can hear a ringing noise in the ear peice to let them know that the other parties phone is ringing. However, when the H323 client rings a SIP client, there is no ringing sound at all, although as soon as the called party picks up the

Hi All I have some grandstream phones registered to my asterisk and all internal, external, voicemail services etc are working very well. I am not sure that it is a problem more so an annoyance. If someone dials my extension number or external DDI while I am already in a call rather than skipping to the next priority in the dial plan for example voicemail the line continues to ring and while in

I'm running * on two pstn lines (x100p cards) that happen to also have analog phones installed on the incoming pair for backup (until testing is complete). If someone is talking on a pstn analog phone and they talk loud enough, * senses the voice (apparently assuming the line is ringing), and rings an appropriate SIP phone (7960). Is there a way to config * so as to only process inbound

I am trying to find an example of how to set up my FXS Station Card in my Asterisk. I have (1) XP100P I have (1) tdm20B (2 Port FXS) Could someone tell me if this is correct? /etc/zaptel.conf fxsks=1 fxoks=2 fxoks=3 loadzone=us defaultzone=us /etc/asterisk/zapata.conf [channels] ; language=en ; ;X100P Port 1 context=inbound-analog signalling=fxs_ks usecallerid=yes echocancel=yes

Hi, I am using Asterisk with various brands and models of SIP phones. Especially the Welltech phones LP201 are particularly nasty with volume and echo. Even with the input gain (microphone) of the Welltech set to the max, the PSTN end can hardly hear the SIP user on incoming calls. Ztmonitor also only gives a level of around 3 === from the SIP phone. I have to increase the rxgain and txgain

I have gotten some cisco VIP 12 and VIP 30 IP phones that I would like to use with asterisk, I have set them up using chan_skinny. The phones work well, except the only problem is that it is like the cisco phones are muted. When I talk on the cisco phones I can hear my self through the ear peice, but the person who I am calling can not hear me at all. I have tried various cisco phones from various

Hey Geoff, thanks for the reply. I had a chance to listen to it for quite a while yesterday and I realized that it's not a constant problem. It starts out really soft, so soft that you wouldn't even notice it if you didn't know what to listen for. I have two types of songs playing at the moment, loud music (rock/punk) and very soft music (middle eastern style). It seems to always

I'm using Asterisk 1.0.9, a Digium TE210P dual T1 card, and two Rhino channel banks (one 12FXO/12FXS, the other 24 FXS). So it's an analog phone on the inside connected to one of the FXS ports, and PSTN line connected to one of the FXO ports. My problem is that as soon as I hear the _first_ ring when I dial out through the PSTN line, I hear a tiny echo on the phone (I estimate between

Sorry if this has been asked before, but does anyone have any pointers on getting the Message Waiting Indicator working on a GS ATA-286? I have tried both settings on the 286's web page for sending a SUBSCRIBE to the SIP server. I was expecting the LED on the 286 to flash when there are messages waiting. I also have a mailbox=XX line in the sip.conf for the 286's extension. Anyone have

Dear group, Was wondering if anyone out there has had the experience I have been having. In reading recent posts on echo cancellation, I think there is.... We recently cut over the Asterisk and are configured with 5 FXS and 2 FXO ports to the PSTN via 2 TDM400P's and 5 SIP phones on our local network. I have set up echo cancellation with 800ms echo training. I do not have

Hi, I've looked in various FAQs and web pages for this info, but just can't seem to find it. When two tones of two different frequencies sound equally loud, what's the (rough) relationship between their power? This is for percussive sounds in music, but I assume it's roughly the same for all sounds. The ear seems more sensitive at high frequencies. For example, when you

I really don't know if this is the same problem that was reported by Ingo Saitz (I really couldn't say which one was 128kbit and which was the original wav when blind-testing. The original had some distortion that perhaps does somehting with my cheapo soundcard) but here's a description of what i've found: When doing some sample encoding with rc2 (rebuilt rpm with latest redhat

Hi All, I've run into a small hiccup in encoding my audios with Speex. When I encode audience laughter and applause with 'speexenc' (version 1.0.1), the result is quite acceptable... until I enable VBR. Then it distorts horribly. My understanding of VBR is that it frees the encoder to vary the number of bits emitted to better maintain the quality requested, and so I would have

Hi, I'm experiencing a few (but meaningful) cases of audio distortion (or bad quality). I can't say yet how often this happens. Please listen to the following sound file: http://213.96.91.201/temp/distorted_audio_1.wav This was recorded by Asterisk while the local SIP caller was dialing out a SIP trunk (so the problem is on my side, definitely, and it doesn't seem to be related to

I have an intermittent problem with the one FXS line that I have. On most calls, the first ~5 seconds of the call has a loud buzzing noise on the line. After 5 seconds or so, it fades off to nothing, and the sound quality is great. Searching for "buzzing" on the list doesn't give a whole lot to work with. The buzzing happens on calls that are routed over both my FXO line and

This is my setup: SPA-2000 -> Asterisk -> X101P (x4) -> PSTN 3-way calling works fine if I use flash and dial just local extensions. Or even if I use flash and dial one local extension, and one remote party over the PSTN. However, as soon as I dial from my SPA-2000 out over the PSTN, and hit flash the call hangs-up about 50% of the time. The other 50% of the time it puts the call on

Dear all, Does any one knows why should I get the following error message, when trying to do a simple data.frame?? DataF<-data.frame(Subject,BiomR,Spp,Capas,Litter,Herbs,LitterD,MaxCanH,DDifS p,DSSp,Slope, CanDens,NearestSp) Erro em data.frame(Subject, BiomR, Spp, Capas, Litter, Herbs, LitterD, : arguments imply differing number of rows: 202, 0 The data I am using

I'm using * 1.0.3 on Gentoo 2004.3, zaprtc from bri-stuff for timing. When I put a caller on hold, the volume of the hold music in the callers ear is extremely loud. I'm using the default entry from the musiconhold.conf: default => quietmp3:/var/lib/asterisk/mohmp3 Volumes with a called or calling party are fine, it's just the hold music volume that seems to be way off

Hi all, We've got a department that has 5 phones using a * 1.0.9 box. They need to have an extension that rings all 5 phones at the same time. Getting all of the phones to ring isn't a problem, but they are running into a problem with the phones ringing in their ears when they are already on a call. Example: Caller one calls the queue, all of the phones rings, and employee one