search for: suppresed

Hello, I'm trying to use Asterisk as a SIP PBX with H.323 trunk connectivity. Everything works except that calls that comes from the H.323 side do not get audio both ways. Since the other way round works fine (calls to H.323 side), I suspect the problem to be in the way VAD or Silence suppresion is negotiated. Is there a way to disable VAD in the Asterisk for H.323 gatekeeper connectivity ? I

I would like to suppress the text 'Scatter Plot Matrix' that appears under the plot. Could someone please suggest how? _____________________________ Professor Michael Kubovy University of Virginia Department of Psychology USPS: P.O.Box 400400 Charlottesville, VA 22904-4400 Parcels: Room 102 Gilmer Hall McCormick Road Charlottesville, VA 22903 Office: B011

I am having some problems with g729 with SIP and ZAP channels. 1) I have two g729 licences. Very frequetnly (I don?t know what triggers the error) I get the following warnings and error when I try to place a call via SIP to my X100P. The only way to get out of this is through a restart of *. When the error ocurrs there are no other calls in place. Any ideas? Error Opening channel:2 not

Dear r-helpers, In basic graphics, I have a figure with x and y axes suppresed. I would like to move the xlab and the ylab closer to the axes. Do I have to use mtext()? _____________________________ Professor Michael Kubovy University of Virginia Department of Psychology USPS: P.O.Box 400400 Charlottesville, VA 22904-4400 Parcels: Room 102 Gilmer Hall...

I hate to bother the list with this potentially minor issue but.... I just wonder if it's a symtom of some other problem. Every time I make a call the BT-102, with the latest firmware, she just keeps the LED display lit and the timer counting after hangup. I check the CLI and the hangup is being executed, I certainly was concerned it might be keeping the line open but that doesn't seem

Hi I have the following setup to make outgoing calls: X-Lite (build 34025) at home behind NAT -> Internet -> Asterisk at work behind NAT -> Internet -> VoIP provider -> GSM gateway -> cellphone. I just tried calling my own cellphone, but there is no sound either way. Here's what I did on the X-Lite at home in the Topology section: IP address : Discover global address

Hello, This is not exactly an Asterisk question, but I was encouraged to seek advice here anyway. The kindness of the * open source community is legendary :) I am getting going with an Asterisk 1.2 box, and I'm having trouble getting good quality transmit sound using handsets with VoIP phones. I'm primarily trying to focus on SPA-941, but also experimenting with Aastra 9113i and Uniden

-- Attempting native bridge of SIP/1009-3df0 and SIP/101-f8f1 -- Attempting native bridge of SIP/1009-3df0 and SIP/101-f8f1 RFC3389: 5 bytes, level 0... Aug 27 08:32:16 NOTICE[23572]: rtp.c:289 process_rfc3389: RFC3389 support incomplete. Turn off on client if possible Killed Whenever I make a call between extension 101 and 1009 which are both Xten Xlite SIP clients, I get that error and

...Even when I explicitly suppress form feeds (via :sf:) I still get the extra stats page. I don't get these extra pages when I print to the HPLJIII or HPLJ4. Does it have anything to do with accounting? I showed the PC/LAN guy how to telnet into the printer, and we gave it a hostname and suppresed the banners. Scratch that; I just checked the printer again and banners were enabled! The guy forgot to save his changes when he exited the printer/telnet connection. I just suppressed the banners, saved the printer config, problem solved. I think I'll post this anyway, just in case it m...

There is experimental echo cancellation code. Look in speex_echo.h and mdf.c. Unfortunately there's no sample code, and I don't think anyone's had much luck getting the echo cancellation to work at all. Would anyone like to prove me wrong? I'd love to see this work! I know it's really hard but just tonight the person I was talking to asked me, "Is there ever

Hello People, I am a newbie asterisk and happy user, i have configured a x100p card and everything works nice, i can forward incoming connections to a x-lite software client and works out of the box, However when i try to make a connection between two x-lite clients then no audio is transmited, i have followed the instructions on voip-info.org, the tutorials on onlamp and i have read some

Hey everyone, I have been trying to figure this out and I am just getting no where with it. The office is using Polycom IP 601 phones. Everything sounds great in terms of quality on both heads. However, users of the phone are having trouble with their headsets and handsets. Some users are hearing their voices back come back on the phone. If they adjust their volume lower, then it goes away,

Hi When i make a call using IAX2, the log of the remote asterisk say Nov 17 20:20:12 NOTICE[28686]: File rtp.c, Line 263 (process_rfc3389): RFC3389 s upport incomplete. Turn off on client if possible Nov 17 20:20:22 NOTICE[28686]: File rtp.c, Line 263 (process_rfc3389): RFC3389 s upport incomplete. Turn off on client if possible Nov 17 20:20:26 NOTICE[28686]: File rtp.c, Line 263

Hi, there: When plot ann predicted results I need to indicate numbers of inputs for each column. V1 V2 V3 V4 1 86.2700 49.9380 30.7630 0.1327 2 89.5127 55.9707 33.7683 0.1186 3 91.1833 58.4670 34.5610 0.1134 matplot(t, pch = 1:4, type = "o", col = rainbow(ncol(t)),xlab = "No. of inputs ", ylab = "Mean of 6 Datasets",

.../log. This means syslog() cannot talk to syslogd and all messages logged by this process are rerouted to the nearest blackhole. The problem is not apparent at the default LogLevel (in fact things look better this way at the first glance because superfluous invocations of auth_log() in the child are suppresed). Some implementations of syslogd are able to listen on additional sockets but 1. other implementations lack this feature, 2. one has to mess with syslogd configuration when sshd is installed. An alternative approach would be to send log messages to the monitor and let the monitor forward them to...