similar to: Unable to open pseudo channel for timing... Sound may be choppy.

Displaying 20 results from an estimated 7000 matches similar to: "Unable to open pseudo channel for timing... Sound may be choppy."

2006 Dec 11
1
Unable to open pseudo channel for timing... Sound may be choppy.
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!
2004 Dec 01
0
Unable to open pseudo channel for timing... Sound may be choppy
Hello, I just sent it with a wrong title... so once again: I just compiled and started Asterisk 1.0.2 following "Getting Started With Asterisk Version 0.1a" from http://www.automated.it/guidetoasterisk.htm I made only one change to default config files - I changed from using oss to alsa. I don't have any devices so far. I started asterisk from the command line: # asterisk -vc
2009 Jan 25
2
Choppy Sound On Bridging From SIP->IAX
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
2007 Sep 06
1
Choppy sound while converting alaw to ulaw
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-
2007 Jan 17
2
One way choppy sound
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
2004 Jan 02
4
one way choppy sound problem !
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
2008 May 05
2
AGI - Choppy Sound
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
2006 Nov 10
1
Choppy sound in voicemail using Asterisk 1.2.11 on CENTOS4 guest on vmware server
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
2009 Sep 27
1
DAHDI Question/Choppy Sound
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
2009 Oct 09
1
choppy sound
Hi After a day of running asterisk, I got choppy sound when fw ip->pstn When I restart asterisk the sound is fine, Anyone had same problem? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091009/be8bbf2f/attachment.htm
2005 Feb 24
1
choppy and cracking sound from zyxel prestige 2002
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
2006 Apr 10
1
Choppy Sound when using linux router or asterisk
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.
2005 May 16
3
Choppy sound
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,
2007 Nov 20
1
Switch to Multi-Proc -> Choppy sound?
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
2003 Dec 27
1
Outgoing call with bad/choppy sound
Hi all. I have this configuration: Telco <-----(E1)----->TE410P//Dual Xeon Server 2.4Ghz<-----(Ethernet)----->Switch<----->GS//BT The Server is running RedHat Linux 8.0 with kernel 2.4.18-14-smp and we are having the following 2 issues: 1.- When making calls from the GrandStream to the PSTN the audio is choopy, plus theres is a pulsing sound, but when the GS
2003 Nov 19
3
RTP timing in a SIP only world (choppy MOH)
I have an * setup with sip phones and sip fxo gateway. When a sip phone places a sip/fxo call on hold, MOH is very choppy. It looks like RTP has a real problem with timing if it is not receiving RTP packets. If the outside call that is placed on hold is not generating any audio, the sip/fxo gateway does not send * RTP packets. Is this valid? Is this a problem with the sip/fxo gateway or a problem
2007 Apr 08
2
intermittent choppy sound over wifi link
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
2006 Mar 17
2
choppy recorded sounds in asterisk
I have installed asterisk on numerous servers. Every install was done on Fedora and (White box Linux). I now have zap channels in one of the boxes (T-1). No matter what type of channel I call on (sip or zap) I get some really strange artifacts in the sound, almost like a skip in the playback. It seems to always be in about the same place in the recording. Usually in the beginning of playback. For
2005 Mar 28
1
Sounds gets choppy after 30 seconds
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
2009 Feb 17
2
Packet Truncated - Choppy Audio
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.