similar to: Sound Quality Problems

Try to use 2.6.10 kernel not 2.6.12.3 kernel Coz I got some problems with the kernel 2.6.12, I had the same Digium Wildcard TE110P, Good luck ________________________________ Da: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Per conto di Robert Christian Inviato: marted? 26 luglio 2005 19.13 A: asterisk-users@lists.digium.com Oggetto:

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, I have discovered i can run starcraft at (almost) full speed on my system. This only works one time, directly after reboot. When tried a second time it's all choppy again. My guess is there're some stale wine processes after the first attempt. When i reboot and start wine instead of winedebug the game video performance is choppy. When i start the game with winedbg after that The

pstn -> sip gw -> * -> C7960 When I dial into * via the pstn, I hear the ivr menu just fine (good quality). I press 3000 (valid extn), and I begin to hear ringing however the ring back is very very choppy. I answer the C7960, and speech is clear in both directions. Place the C7960 extn on hold, and the MOH is very choppy. Checking 'sip show channels' indicates both the sip gw

Hi All, I've just setup a test Asterisk system that allows incoming/outgoing calls via an ISDN card (l4i) and incoming/outgoing calls via SIP (iconnecthere). I have two SIP Softphones (Xten X-Lite) for making and receiving calls. When receiving an incoming call via the ISDN interface the sound quality is fine for the Softphone user (i can hear the caller perfectly), but the person

Hi, I've been having issues with asterisk playing back recorded messages. They sound clear..but there are lots of breaks during playback (like its losing packets). I got top-end hardware and I'm on a killer network so its not that. I've talked over my SIP line using a regular telephone and it sounds great, so its not the VOIP provider. Asterisk is working great with no other

I have a client with about 16 GXP-2000. They complain that the audio quality is terrible after 2 or 3 simultaneous conversations. They are behind DSL 1.5Mbps down and 256Kbps up. Because they are using G711.u codec, I know they upstream bandwidth is the limiting factor and they most likely won't be able to have more than 3 simultaneous conversations, and if they're surfing the

Any users of the VocTel VOIP service? (Canadian) How have you found the quality (Choppy / smooth audio)? Any problems registering? (I have been unable to register for hours) After reading about the collapse of a big USA VOIP provider, I'm curious Thanks, OCG -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I have asterisk-0.4.0 running. When I make a call between an ATA186 and Asterisk using ulaw or alaw codec, all is fine. I installed the CVS version and tried the same thing but the voice is choppy. The installation was done on the same linux server. The stats on the ATA186 show no packet loss but a great number of "late packets". The stats when running version *-.0.4.0 do not

Hi All, I'm running Asterisk on the following vendor_id : GenuineIntel model name : Celeron (Coppermine) cpu MHz : 668.202 cache size : 128 KB with 192 MB Ram Audio coming from Asterisk (the demo ) is excellent when using a SIP phone on the LAN to Asterisk, and when dialling in from outside via ISDN to Asterisk. However, when connecting from SIP phone to SIP

Greetings! I have made great progress thanks to this group. My Asterisk seems to be working for the most part. I am using the following equipment/software: * HP Vectra VL - Pentium Pro CPU - 256MB RAM * Redhat Linux 8 - Loaded straight from distro CDs as Developer Workstation - latest updates from RHN * Asterisk (latest as of two weeks ago when I used CVS checkout) * X-Lite SIP Client on a

Hardware: Digium TE110P REV.C and REV.D Gigabyte GA-965G-DS3 Bios F8b cat /proc/cpuinfo .... model name : Intel(R) Core(TM)2 CPU 6600 @ 2.40GHz stepping : 6 cpu MHz : 2400.080 cache size : 4096 KB ... latest libpri, dahdi, asterisk as of tonight. linux: debian lenny After moving hardware around all slots, disabling all unused hardware with no

60-cycle hum ?? yep, probably a ground loop. http://www.xiph.org/archives/icecast/2703.html -----Original Message----- From: Scott Prive [mailto:Scott.Prive@storigen.com] Sent: Monday, May 20, 2002 10:04 AM To: icecast@xiph.org Cc: bikepunx@impop.bellatlantic.net Subject: RE: [icecast] soundblaster live noise <p>I don't know the answer, but here are suggestions to the limit of my

I'm running Asterisk 1.2.13 on a Solaris 10 X86 box behind an IPCop firewall on a 5Mbps down/512 up cable connection. I'm having sound quality problems when users call in for voicemail and with music on hold. The sound is choppy and muffled while souding pretty good for calls inside the network. I'd appreciate some pointers as to where to start looking to improve things. I've

I have really tried to do my best googling and wiki-reading before asking this question. I couldn't find the answers there so I throw myself at the mercy of the list... I get excellent quality for SIP -> PSTN and PSTN -> SIP calls, however when I or anyone else calls from PSTN -> * the voice menus are oftentimes very choppy. Sometimes they are absolutely perfect and I cannot tell

Greetings! I have a FreeBSD 5.3 running on Intel SR1300 (dual xeon 2.6, scsi) server, with ztdummy.ko driver as timing source for asterisk. The typical output from zttest is: $ zttest Opened pseudo zap interface, measuring accuracy... [..skip..] --- Results after 192 passes --- Best: 99.987793 -- Worst: 98.266602 But when i listening MOH, it's quality is not very good. It is slightly

Hi, I have the VIA chipset, and I'm trying to disable the sound and enable a soundblaster compatible card. Can you tell me what you did in /etc/modules.conf to enable your soundblaster card? Thanks, Mike

Forwarding this to wine-users, I made a mistake while sending. Gardou J?r?me <jgardou at yahoo.fr> a ?crit : Date: Fri, 14 Mar 2008 14:09:42 +0100 (CET) De: Gardou J?r?me <jgardou at yahoo.fr> Objet: RE : [Wine] getting a Creative Soundblaster card to work ?: Susan Cragin <susancragin at earthlink.net> Cc: wine-user at winehq.org Susan Cragin <susancragin at

I wasn't thinking of using the conference system as the basis. I was thinking more along the lines of: 1) Setup a second extension on the Cisco phone named "INTERCOM" enabled for auto-answer 2) Create a call group on asterisk to dial that "INTERCOM" extension on every phone that will participate 3) Add a feature code that would dial the intercom extension and connect

Hi I've problem with incoming call quality to GSM gateway connected to beronet card (BN8S0), -----> [ GSM Gateway ] -------> [ BN8S0 ] ==== asterisk Port connected to GSM gatway is in TE mode , gateway is in NT mode , When I dialin to cellphone numer , call goes to 'from-eragsm' context, to Echo application. [from-eragsm] exten => 700,1,Goto(600,1) exten