[Asterisk-Users] IVR Menu and VoiceMail quality

[Chris Shaw]

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...

----- 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
> > > >
> > > > _______________________________________________
> > > > Asterisk-Users mailing list
> > > > Asterisk-Users at lists.digium.com
> > > > http://lists.digium.com/mailman/listinfo/aster> isk-users
> > > > To
> > > > UNSUBSCRIBE or update options visit:
> > > >
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> > >
> > > _______________________________________________
> > > Asterisk-Users mailing list
> > > Asterisk-Users at lists.digium.com
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > > To UNSUBSCRIBE or update options visit:
> > >    http://lists.digium.com/mailman/listinfo/asterisk-users
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/aster> isk-users
> > To
> > UNSUBSCRIBE or update options visit:
> >
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users