similar to: Choppy communication issue

Hello! I'm new to Asterisk, although I've had my eye on it for about a year now. I just recently installed in on RedHat 8 on a 2 GHZ system, but the sound was choppy - presumably from the onboard sound card (I read about that in the archives). So I stuck Asterisk on an old ISA system - 300 Mhz, 100 mb ram, RedHat 8, and SoundBlaster 32. While there was much improvement in the

Hi all, I want to move a filesystem off of my NT server over to my Network Appliance NFS server to then be shared via Samba. I assume the easiest way to accomplish this would be to use smbtar. Do I then use smbtar to extract the tar file I create with smbtar, or, can I use gnutar? Does it matter? Am I safer sticking with smbtar? I'm guessing that since smbtar is a wrapper around

Hi all, I'm using Samba 1.9.18p8 under Solaris 2.5.1. Within my smb.conf file I am using the 'netbios aliases' option to make my server appear as 4 different servers with an smb.conf.%L to load the specifics of each: netbios aliases = hardware software diagnostics homedirs include = /usr/local/samba/var/smb.conf.%L The problem I'm encountering is that the only comment I see

Hi all, I've recently set up a samba server, but just discovered that people on the "other" side can't reach it. The system shows up in Network Neighborhood, but you can't double click on it. It returns a message saying that the server is either inaccessible, or that I misspelled the name (How I did that with just one mouse click is beyond me ;) I had 'allow

Hi all, I've been running a Samba server here for over 6 months now will great success. This past weekend I moved the Samba server to a new subnet, and now I can't see any of the various aliases from Samba server. I was previously able to see the aliases in NN across my 2 different subnets by using the 'wins server = ' parameter. The wins server I was registering with is on yet

OK, background/config. running * (show version reports 0.9.0) on Mandrake 9.2 (kernel: 2.4.22-32mdk) with a dual 800mhz PIII with 256M Ram 4port FXO digium card, no IRQ sharing I can find (cat /proc/pci & cat /proc/interrupts), vmstat reports a minimum of 80+% CPU idle when problem occurs. connect to a Grandstream 101 (GS) via vpn (no nat). Link has 100ms - 150ms ROUND TRIP latency

If you don't want to type the ip adress of your samba server every time you connect, add a entry to your local hosts file and lmhosts file. You will find this files (and sample files with extension ".sam") in the windows main directory of Win95 and under "system32/drivers/etc" on WinNT 4.0. Regards, System-Consulting Kollien Rudolf Kollien Email: kollien@kollien.de Our

I have just installed the Alsa drivers for my 2.4.18-14 kernel (RH8). I have configured the sound card ok with alsaconf and tested with the aplay , works fine. But when I run asterisk it says.. ------------------------------- [chan_alsa.so] => (ALSA Console Channel Driver) Apr 20 18:28:34 ERROR[8192]: chan_alsa.c:339 alsa_card_init: snd_pcm_open failed: No such device or address Apr 20

Hi, I'm having a problem with DTMF tones from my SIP client apparently crashing the chan_capi driver. However I'm not sure whether this is a bug or misconfiguration on my part: if I set "softdtmf=1" in /etc/asterisk/capi.conf the problem goes away. Does the AVM B1 not support DTMF detection? The set up I have is using latest CVS (3 days old) running RH8 on a 933MHz P3. SIP

Hi all, I'm currently battling with my Corp. IS group over NT domain administration. I admin a mostly Unix environment with a lot of WinXX clients. The authentication server is owned by "Corporate" (you know, that nameless, faceless entity that always makes do stupid things because "That's the policy" :) and I have no control over it. What I'd like to do is

I am experiencing choppy sound when I bridge from a SIP peer to an IAX peer. I am running Asterisk 1.4.13 on a 2.6.22.9 kernel (Fedora). I am experiencing choppy sound from the SIP peer to the IAX peer but not vice-versa. I know that this is not a bandwidth issue because I don't have choppy sound (with the same codec) when bridging IAX->IAX peers or SIP->SIP peers. My timing source is

Hi there I europe alaw is usual. I have a SIP Phone which perferes ulaw. When my * box has to transcode alaw to ulaw the sound get's one way choppy. (alaw => ulaw is choppy, ulaw => alaw is fine). I managed to fix the issue by forcing my SIP phone to use alaw only, but is this a know issue with asterisk 1.2.13? -Benoit-

Hi there, We're having some complaints of choppy audio from our SIP customers. Asterisk is showing no errors, but I'm getting a lot of these in my syslog: Feb 17 13:34:31 ntop[2863]: **WARNING** packet truncated (14654->8232) The first number varies, but the last number is always 8232. I've read that this is a common MTU size, but none of our interfaces have an MTU of 8232.

Hello list, I've been experiencing choppy sound as well. The version on Asterisk I was using originally was dated 10/24/03 (I think), the problem appeared after I updated from that version. My setup is a little different though. I'm having choppy sound only on some incoming calls -- from PSTN->PBX (between spans on a TE410) and PSTN->SIP. We use Cisco 7940 handsets and we also

I have had no success in getting the voicemail working on Asterisk 1.2.11 on CENTOS4(2.6 kernel) guest on vmware server 1.0.1. I tried with or without ztdummy device, renice -20 on asterisk process and even real-time priority on the host Windows XP box for the vmware process. I am running on an AMD Athlon 64 X2 4600+. The behaviour is when the voicemail answer, the voice sound ok but when

Hi Guys I'm conecting 2 astersk servers using this arquitecture (Ext softphone)<==sip==>(asterisk 1)<====iax2 trunk====>(asterisk 2) <===alaw==>(pstn) If i call from the Ext to the asterisk 2 the sound is perfect, but if i call from Ext to the pstn, i can hear perfect but they tell me that sound really choppy, i tried using several codecs (same problem) but i

Hi! I have Asterisk 1.6.1 installed on OpenSuSE 11.0 running with choppy sound. One specialist on the forums asked me if I have DAHDI configured, he assumed that this could be cause of choppy sound problem. > dahdi_test Unable to open dahdi interface: No such file or directory Do I need to configure DAHDI even if I do not have any Zaptel devices? Is there any guide for configuring

Hey. I'm new to Linux based operating systems. A friend of mine introduced me a year ago and I finally decided to try it so try to bear with me here. The problem I've been having is that I've managed to get Amnesia to work under the latest Wine version (1.3.31) but the game is really choppy. I've tried messing with the Wine configuration and the in-game configuration but it

Hi All, Whenever a call comes in via the ISDN and somebody leaves a voicemail, the sound file recorded is very choppy. If I actually take the call, the sound is not choppy so it's obviously something to do with the Asterisk box itself having to do the recording. Perhaps the sound card drivers? I'm using the stock i810_audio (OSS) drivers on Fedora Core 1. If I call from a local VoIP

Hello All, We recently changed our asterisk system to begin using G.729a as the primary codec. We have a Cisco 1700-series router which connects to the PSTN via FXO ports, along with Cisco 7940 SIP phones. Everything is working great, except... When an inbound caller calls into our system, they hear an IVR. When the caller dials an ext (SIP phone), the ringing progress tone is