similar to: choppy sound

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 folks, I'm experiencing some problems with sound through phpAGI ... What I'm trying to do is a menu, doing some database lookups and so ... But sometimes the sound become too choppy ... just sometimes .. like 1 of 5 calls ... but is a big percentage ... And I have my current menu on the dialplan that I have no problems with it ... I'm using .gsm for both but different

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! 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

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 all, I have my asterisk setup as following: IP 2 x E1 x-lite <-------> Asterisk -------> PSTN When I place a call from x-lite to PSTN, the quality of the sound in the direction x-lite -> PSTN is very bad. That is, the voice of the x-lite user, heard by the PSTN user is choppy and makes communication not very pleasant. The sound is choppy as if bits of data

Hi, Does anyone have suggestions hooking Zyxel Prestige 2002 to Asterisk? I have tested Zyxel Prestige with both supported codecs. Call with G.711 sounds very choppy and cracking. Almost can't understand a word. Today I installed G.729 support into Asterisk but unbearable voice quality remains. It's a little bit better though. I have tested that Zyxel ATA with some commercial SIP

1.) I'm getting messages in my log: Unable to open pseudo channel for timing... Sound may be choppy. Unable to open IAX timing interface: No such file or directory I'm using kernel 2.6, I don't think I timing, do I? 2.) I'm losing IAX registration with provider nor IAX protocol will go through. Though, I can ping both providers just fine. When I reboot the firewall, the

Any idea what causes the warning "Unable to open pseudo channel for timing... Sound may be choppy."? Any ideas what I need to resolve this? I do have the zaptel module installed but don't have a zaptel card. I'm guessing this has to do with ztdummy? I'm running Debian and installed asterisk, zaptel, and zaptel-source from the backports. Any information appreciated!

Hello, I created this setup, DSL------LINUX ROUTER-------ASTERISK Linux acts as router and forwards packets only 512M and AMD 1599.987 MHz Asterisk 512M AMD 2000 MHz When I ssh to linux router during the call and execute any command that requires cpu , then sound gets choppy. Simple test would be establish a call and start "du /" on the router. The same applies to asterisk box.

Hello all. I have a strange and irritating problem with Star Wars: Knights of the Old Republic. I played ok up to a place where I saved. There, I quit the game. When I restarted, the sound became choppy, making the game unplayable: about 1/2 sec of sound plays, then loops for about 3-4 sec, then the next 1/2 sec, loops again, etc... The screen isn't updated during this; well, it is updated,

Hello, everyone I'm relatively new to Asterisk (and VOIP in general), but I have a project that it will really help with. So, I setup a test system on an ancient 400MHz P3 we had lying around. It worked great. I had a test dialplan working, and had no trouble connecting to it with SIP using 3CX SoftPhone over our LAN (and over the Net through our NAT). So, we went ahead and bought a

I am experiencing a situation where I am getting intermittent choppy audio. Here is the network layout: Termination provider -> IAX2 over the Internet -> 20Mb fiber connection -> router -> Asterisk My ATA connection goes into the router between the fiber and the Asterisk server on another interface here is the layout from me to Asterisk: Sipura ATA (SPA1001 running

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

Andrew, Please find links below to sound clips from different combination. You will see that the 2.6.9.34 kernel on trixbox 2beta still does not sound ok. What did you do on the trixbox 1.2.2 vmware load to make it work better? I say better because it is still not perfect as you will notice that it accelerates near the end. I really appreciate your help and the great work you are doing. Also,

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

This is driving me crazy, when making an outgoing call the first 30 seconds is always perfect, then the party on the receiving end can always hear me perfectly but after 30-60 seconds the audio coming back to me from them starts to get choppy and drops out. I've tried this with multiple devices, from multiple locations some behind NAT, others not. This is using the ulaw codec, although

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.

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