similar to: Software echo with Jack module

Hello everyone! So I hacked app_jack.c today, as best I could. Whic came mostly down to inserting ast_log() messages. I discovered the following with JACK: When it starts, it tries to read 512 bytes and only gets 0. That clears up after a while. Sometimes a good time later than the reading comes the writing. And there the real strangeness might begin. Because usually the framebuffer

Hi, I want to use asterisk with jack audio. I tried the next configurations : 1 ) app_jack.so -> does not work. 2 ) chan_alsa.so without jackd started my .asoundrc : pcm.!default { type hw card 1 } pcm.jack1 { type jack playback_ports { 0 alsa_pcm:playback_1 1 alsa_pcm:playback_2 } capture_ports { 0 alsa_pcm:capture_1 1

Hi, I have been headbanging with asterisk and Jack for a while, decited to ask other linuxists for an advice. The problem is that Jack is compiled from source (0.118) in /usr/local/, but menuselect says "XXX" for it (cannot enable it). I need jack... Otherwise I will inotify Monitor WAVs, what is bad :`( After installing jack from sources: Add system-wide

> > Different brand/model, but similar as they are both el cheapo, > entry-level headsets. I tried using them on a laptop, and I get > marginally better microphone output, even with its volume cranked all > the way up + automatic gain control enabled. > > I guess those on-board soundcards by Realtek aren't as good as a > quality microphones. I'll get a USB headset

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I'm running the latest firmware on the SPA-841 and have a problem with echo. The echo occurs on all calls (PRI ISDN on a E110p or SIP) and is not present when I use the SNOM190 phones so I can def. isolate it down to the SPA-841s. The codec used is g711u and the phones are on their own dedicated 100mbit switch with no other traffic. The server is a 3Ghz PIV sitting at 99.9% idle all

Hello, I'm having the following problem when using a headset on XP connected to an on-board Realtek soundcard on an AsusTek M2N68-AM Plus motherboard: - Using any sound recorder (Windows', Audacity, XLite), the level is just too low when speaking at a conversational level, even with the microphone level pumped all the way up (line displayed totally flat in Recorder)

Hi, On Tue, Aug 24, 2004 at 10:44:49AM -0700, Shana Cooke (Gitnick) wrote: > The result has my husband speaking clearly, but the echo of myself > talking is garbled and sounds like the Borg Hive from Star Trek. It's > more distracting than the original signal was. I have tried introducing > delays to my echo, but it does not help the signal to become clearer. Try this patch. It

I'm hearing an echo when using a headset with my IP 501 / 601. The phones are using BR 3.1.2 and SIP 1.6.3. I use tftp to configure the phones. The sip.cfg is the default from polycom except for the parameters required to connect the phones to asterisk. I have absolutly no echo with the handset, but do have a slight echo on the speaker phone. I haven't ruled out room acoustics as the

Good morning, I just got asterisk up and going and set it up to communicate to test calling out to a PSTN phone number. My configuration is as follows: My Diax client --> IAX2 on my asterisk server --> IAX2 (voice pulse server) --> PSTN -->my PSTN phone. I don't have any ZAP hardware. I'm only communicating using IP. For timing, I installed the ztdummy module successfully

Thanks J. It didn't work. On 05/10/2017 01:57 PM, J Montoya or A J Stiles wrote: > On Wednesday 10 May 2017, andre castro wrote: >> Hello, >> I am new to Asterisk, so please bear with me. >> I have made a success installation from source of Asterisk 14.4.0 on >> Debian Jessie (8.7). And I am running the Asterisk server, with several >> extensions and

Hello I just set up Asterisk 1.6.2.9 through packages on a test host running Ubuntu 11.04, configured sip.conf/extensions.conf, and launched EyeBeam 1.5.20 to run the echo test. For some reason, even through I'm using a headset, there's a lot of echo and after a few seconds, it sounds like it enters a very fast loop before the echo stops somewhat. IOW, unusable sound. Here's a

I recently installed * on my firewall and that of a relative some miles away. I route sipphone (kphone and x-lite) calls from deep within the backbone (two layers of firewall) on each end to the other. Works fine between @200Mhz pentium doorstop linux boxes (even w/2.4 kernel).The problem of course is the output of the speaker at the other end is picked up by the microphone (confirmed by

I have Asterisk running on my LAN with softphone clients (SJPhone) and Cisco 7940/60s, all using SIP. I also have a few remote sites connecting to my Asterisk server. I am getting an echo back of my voice when talking with one particular site. The caller does not hear an echo on their end. All calls on the LAN or to other sites do not produce an echo. When the caller places his SJPhone on

Hi, I am currently working on a thin-client "low-cpu" usage telephone application. The current setup is that you can make phonecalls using an alsa supported usb headset and a smartlink based usb modem (a sweex usb modem, which -in large quantities- goes for 17.95 euro's) for analog lines. Unfortunately, it is really made as a modem and not for audio purposes, so, there is a pretty

Hello, I am new to Asterisk, so please bear with me. I have made a success installation from source of Asterisk 14.4.0 on Debian Jessie (8.7). And I am running the Asterisk server, with several extensions and dialplans, all working well. However I am struggling to get app_jack to run. In menuselect I can see that it is XXX due to dependencies on jack and resample, however both Debian packages:

While I'm asking about the Polycom ip500, the answers for all phones where mic/handset/headset levels are adjustable would be of interest to many I'm sure. For the ip500, the default value for the handset seems to be voice.gain.tx.analog.handset="3" I've noticed that echo all but goes away when one reduces the mic volume on almost any phone. My question is, for you users

Hello everyone! So now I'm testing with chan_sip and I discovered, that I can make calls, even if they're only listed as active channels. But JACK just emmits white noise, with a highger frequency than 8kHz, in my believe. A call with app_record shows, that the signal is clear and very ear friendly though. So how to proceed from here? What can I do to help someone look into

Hello everyone! So I've just scanned through the debug log, defined like this in logger.conf: full => notice,warning,error,debug,verbose I couldn't see any reason for the connection not working. I called my friend, he heard ringing, accepted the call and then it got hungup. I didn't see any output from app_jack though. Any idea, how I can get more output from app_jack?

I am trying to create a new variable which uses the suburb names if HR and HRRES are the same but which uses HRRES if they are different. Any assistance would be appreciated as my brain has just packed up. I'm not sure I can teach myself anymore new tricks this afternoon. HR HRRES SUBURB What I am trying to get 954 Wheatbelt