[Asterisk-Users] IVR Menu and VoiceMail quality

matt.riddell at sineapps.com matt.riddell at sineapps.com
Sat Jul 10 06:21:29 MST 2004


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" <jay at skimmilk.net>
> To: <asterisk-users at lists.digium.com>
> 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:chriss at watertech.com]
> > > Sent: Friday, July 09, 2004 3:11 PM
> > > To: asterisk-users at lists.digium.com
> > > Subject: Re: [Asterisk-Users] IVR Menu and VoiceMail quality
> > >
> > >
> > > I thought it was only needed for MeetMe and MOH?
> > > ----- Original Message -----
> > > From: "Jay Milk" <jay at skimmilk.net>
> > > To: <asterisk-users at lists.digium.com>
> > > 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:chriss at watertech.com]
> > > > > Sent: Friday, July 09, 2004 1:14 PM
> > > > > To: asterisk-users at lists.digium.com
> > > > > 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: chriss at watertech.com
> > > > >



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