On 9 Jul 2004 at 14:08, Chris Shaw wrote: > Thx Jay, I hope this is not a too FAQ... I really did try to look it up > first but I saw soooo many conflicting things about timing... one person > says no you absolutely do not need ztdummy or a digium card to make > IVR/Voicemail work, others say you need it for everything... I tend to > believe the latter since it seems to be more of a timing issue than a > bandwidth issue... > > What I can't figure out though is if it's a timing thing, shouldn't calls on > my local net be crappy too? When I log into voicemail from my ip phone it's > perfect... when I call home from out of town it sounds like crap unless I > play with the nice values or restart asterisk...
Just a thought, when setting up your QOS, did you make sure that the maximum usage was slightly below your actual pipe size? Matt > ----- Original Message ----- > From: "Jay Milk" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Sent: Friday, July 09, 2004 1:48 PM > Subject: RE: [Asterisk-Users] IVR Menu and VoiceMail quality > > > > AFAIK, it's needed anytime asterisk streams audio... Which is meetme, > > MOH and of course voicemail and IVR. My Asterisk system had lousy IVR > > quality until I plugged in the FXO card and loaded Zaptel. > > > > > -----Original Message----- > > > From: Chris Shaw [mailto:[EMAIL PROTECTED] > > > Sent: Friday, July 09, 2004 3:11 PM > > > To: [EMAIL PROTECTED] > > > Subject: Re: [Asterisk-Users] IVR Menu and VoiceMail quality > > > > > > > > > I thought it was only needed for MeetMe and MOH? > > > ----- Original Message ----- > > > From: "Jay Milk" <[EMAIL PROTECTED]> > > > To: <[EMAIL PROTECTED]> > > > Sent: Friday, July 09, 2004 12:21 PM > > > Subject: RE: [Asterisk-Users] IVR Menu and VoiceMail quality > > > > > > > > > > Do you have ztdummy loaded? > > > > > > > > > -----Original Message----- > > > > > From: Chris Shaw [mailto:[EMAIL PROTECTED] > > > > > Sent: Friday, July 09, 2004 1:14 PM > > > > > To: [EMAIL PROTECTED] > > > > > Subject: [Asterisk-Users] IVR Menu and VoiceMail quality > > > > > > > > > > > > > > > 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 that it's actually VoIP. > > > > > Sometimes it's so bad that I can't understand what Allison's > > > > > saying at all... Calls on the same network sound just fine... > > > > > I know what you're thinking, it's a congested link, and that > > > > > may be but I've noticed that if I play with the nice value of > > > > > asterisk, it seems to help. Setting nice to 0 seems to work > > > > > the best, I tried -20 and it was the worst... > > > > > > > > > > I have implemented QoS on my network and have given any and > > > > > all asterisk packets priority. As I said actual calls are > > > > > crystal clear so I believe it to be a problem with Asterisk > > > > > itself or the machine it's running on. Possibly some > > > > > bottleneck somewhere. I realize that since it's going over > > > > > the public internet, the occasional dropped packet is to be > > > > > expected, but what's frusterating is that I can get crystal > > > > > clear menus sometimes even when my network is fully loaded > > > > > and other times when it's perfectly quiet it sounds > > > > > absolutely horrible... > > > > > > > > > > Here are the machine's specs if that helps: > > > > > > > > > > AMD Athlon 1Ghz (Old Thunderbird core) > > > > > Asus A7V600 > > > > > 128MB DDR-266 RAM > > > > > 450GB storage (4 IDE drives in an LVM array) *grin* > > > > > Pure VoIP, no digium hardware > > > > > > > > > > Internet connection is cable with 3Mbit downlink and 256Kbit > > > > > uplink... > > > > > > > > > > As I said earlier I wouldn't have even asked, but it dosen't > > > > > seem to be totally bandwidth related so I'm wondering if > > > > > anyone has any ideas... > > > > > > > > > > Chris Shaw > > > > > IS Manager > > > > > Water Tech Industries > > > > > Phone: (888)-254-8412 > > > > > Fax: (503)-261-9118 > > > > > E-Mail: [EMAIL PROTECTED] > > > > > _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users