[Asterisk-Users] IVR Menu and VoiceMail quality

Sun Jul 11 12:47:11 MST 2004

I'm really glad you found it useful, I'm just trying to do my part since you
guys have been very helpful to me. Asterisk has been soo cool, I just want
to make sure everyone thinks so :)

    -Chris
----- Original Message -----
From: "usedcanon" <usedcanon at yahoo.co.uk>
To: <asterisk-users at lists.digium.com>
Sent: Sunday, July 11, 2004 12:40 AM
Subject: RE: [Asterisk-Users] IVR Menu and VoiceMail quality

> -----Original Message-----
> From: asterisk-users-admin at lists.digium.com
> [mailto:asterisk-users-admin at lists.digium.com]On Behalf Of Chris Shaw
> Sent: 11 July 2004 08:35
> To: asterisk-users at lists.digium.com
> Subject: Re: [Asterisk-Users] IVR Menu and VoiceMail quality
>
>
> Yep I sure did, damn upstream pipe gets so congested I had to drop it to
> about 75% to keep from dropping packets... Seems to be working
excellently,
> I tried downloading a large file and doing some interactive SSH with no no
> noticeable degradation... I'd say we have a winner. Installing and running
> Ztdummy seems to have done the trick, I cannot tell a difference between
the
> quality over VoIP and POTS now, it's excellent...
>
> So for anyone confused on this issue, if you run a pure VoIP setup with no
> digium hardware and you want asterisk to do ANYTHING, not just MOH and
> MeetME you MUST have some kind of timing source, either ZTDummy or ZapRTC
> installed. Especially for doing VoiceMail, that seemed to be the worst for
> some reason...
>
> This was very confusing for me because the wiki says that it's only for
MOH
> and MeetME, that's simply not true or at least not in my experience.
>
> ----- Original Message -----
> From: <matt.riddell at sineapps.com>
> To: <asterisk-users at lists.digium.com>
> Sent: Saturday, July 10, 2004 6:21 AM
> Subject: Re: [Asterisk-Users] IVR Menu and VoiceMail quality
>
>
> > 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
> > > > > > >
> > _______________________________________________
>
> Your experience has been very useful for me, thanks for sharing it with
> others. I wish more people did this, I mean updating the list after they
> find a solution.
>
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