[Asterisk-Users] IVR Menu and VoiceMail quality
usedcanon
usedcanon at yahoo.co.uk
Sun Jul 11 00:40:53 MST 2004
-----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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