similar to: Low RX volume and half duplex/"walkie-talkie" on AEX-804E

We've had an issue since we went live nearly two years ago on Asterisk where people complain about not being able to talk while someone else is talking. I had assumed for a very long time this was because of the phones we went live with (Grandstream GXP-2000's) and for the longest time I believed this was a speakerphone problem only. Last week during budgets, a request to buy new

Dears; I need to order Digium card and not able to know which one is the best quality? Is it that of AEX with the end E or EF or P or B? I saw those card that its slot is small (I think those that end by EF), are they the best card? Really I am caring to have a card that has echo cancelation and the voice volume is high enough (because previously I faced the problem that the voice volume was

Anyone know if it is possible to control how aggressively the "Aggressive" mode behaves. Meaning, is it possible to dial back the aggressive mode to have a happy medium between Regular and the Aggressive defaults. I have a situation where Normal echo cancellation is not quite enough, however when I turn on aggressive mode We are attacking it to hard and I am unhappy with the walkie

Hey sorry to hijack this thread, but I just remembered a request I wanted to make to the speex devs. I tried using the activity detector, but I just couldn't get it working well. I ended up using my own, where I think it just considered voice on if it passed a certain threshold (I know, pretty primitive). I also tried one that checked for a signal, like if the strongest frequency

Hi guys, I'm building a server that need to host 2 digium TDM24 cards. I know any 3U server with 2 PCI-E slots would do. Since I do prefer supermicro server, but getting one configured is pretty darn hard. Any suggestions here? Cheers, Kelvin Chan | Positronics Ent. Product Development | | unit 272 604-628-9330 (direct) | 8128 128th St.

Hi, I am trying to make a program in C to communicate with a temperature sensor over my serial port that uses RS-485 and is a half duplex. I am however having trouble getting the device into RS-485 mode, is it possible to use RS-485 mode in Centos 6.5 or does it only support RS-232 mode?

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Greetings everyone, I've been having some strange issues with my Asterisk box and some snom phones. In some cases, when I talk, the sound in the other end is cut off, I stop earing the background noise - looks like a walkie-talkie. I've tried this between phones in the same network and in all but one this happens. The one where it doesn't happen is the one connected directly to the

> Anyway, my request is, can you build in a pre and post buffer into the > VAD? In mine, if I detect voice any time between now and say a quarter > second later, I start sending, and then I wait a half second or whatever > after I stop detecting. You pretty much have to have this, or people > start getting anxious talking over an internet stream. They have to > enunciate

Hi All, We've been running Asterisk for years in our offices but just recently replaced an Asterisk Appliance* in our smaller office with an actual server, upgraded the server in hardware in our HQ location and upgrading both ends to 11.5.0 with Gareth's patch for Cisco phones. 99.99% of our endpoints are Cisco 7961Gs. Each office is more-or-less standalone for ease of management and

Hi All, I've looked through the archives and tried several variations in Google, and I haven't found anything on-point... So I'm hoping someone here may be able to help this relative Asterisk neophyte shed some light on an issue: I have a box running Asterisk 1.4.22 in our lab with several Cisco 7961G phones and an AEX804E card (4 FXO, hardware echo cancellation). The server and all

This must be a simple issue, but I cannot figure it out. I want to use VAD, but I don't know how to check if the actual frame has voice in it or not. So, in my code, I do: int tmp = 1; speex_preprocess_ctl(preprocess_state, SPEEX_PREPROCESS_SET_VAD, &tmp); speex_preprocess_ctl(preprocess_state, SPEEX_PREPROCESS_SET_DENOISE, &tmp); then later, for each frame

Hi All, Got a strange (at least IMHO) issue that doesn't make much sense to me. Basic configuration is two sites with a site-to-site (aka router-to-router) VPN. Both sites have Cisco 7961G phones [with SIP firmware] on users' desks, and the only VoIP is internal - all of our outward telecom is on POTS or Centrex-enabled POTS lines. Site 1 has a Dell PowerEdge 1950 with Asterisk built

I am hoping to replace my Nortel 8x24 switch with Asterisk. Right now my cabling comes from my outside phone box into my office and into a punchdown block and leaves the punchdown block as an amphenol connector which currently plugs into the Nortel swicth. A second amphenol connector them plugs into the switch and extends to another punchdown block that I believe carries the lines throughout the

I've been given the task to hunt down a codec that out solar car team can use to transmit and receive data over a radio link between a chase car and the solar car its self. As of now I do not know the exact specifications of the internal hardware of the car, what I do know is they are looking at transmitting across a 2400baud connection, the unit in the car is homemade from our labs here.

Hi, Most companies we work with, have 'designated' crisis management teams. These vary from the insignificant crisis', through to life-threatening crisis'. There is always an assigned emergency services contact, whose job it is in an emergency, to maintain communication with the emergency services. One of our corporate functions is crisis management - so we have to consider

Well, to do an upgrade on a traditional system you have the same issues, perhaps even worse as everything is physically wired to one system. To develop for production you must have a dev environment, a beta test and a scheduled release right? Todd Jonathan Moore <moorejon@usd465.com> wrote: __________ >These are good issues, but I am even thinking of something simpler and more

A month or two ago I sent an email to this list proposing to expand the list of "standard" tags for Ogg Vorbis. No tag would be required, but if you wanted to encode certain types of information about a file, you could use a standard tag. I went through the whole discussion, and revised my proposal in light of all the comments from everyone. Here is the updated proposal. This

Hi All, I posted this a couple weeks ago with no response, I'm hoping that someone will see it this time around and be so kind as to offer advice for resolving this issue (or point me in the direction of a better place to ask) "Some" (but not all) calls on one of our Asterisk boxes are being dropped in Voicemail -- only in voicemail -- after about 20 seconds with the error logged

https://bugs.freedesktop.org/show_bug.cgi?id=102722 Bug ID: 102722 Summary: [regression] changing resolution hangs the display on Lenovo Thinkpad p51 Product: xorg Version: unspecified Hardware: x86-64 (AMD64) OS: Linux (All) Status: NEW Severity: critical Priority: medium