Re: [Asterisk-Users] IVR Menu and VoiceMail quality
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" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Sunday, July 11, 2004 12:40 AM Subject: RE: [Asterisk-Users] IVR Menu and VoiceMail quality > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Chris Shaw > Sent: 11 July 2004 08:35 > To: [EMAIL PROTECTED] > 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: <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > 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 s 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-rea
RE: [Asterisk-Users] IVR Menu and VoiceMail quality
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Chris Shaw Sent: 11 July 2004 08:35 To: [EMAIL PROTECTED] 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: <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> 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 s 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
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: <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> 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 s 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 w
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 s 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
Re: [Asterisk-Users] IVR Menu and VoiceMail quality
Thx Jay, I hope this is not a too FAQ... I really did try to look it up first but I saw s 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" <[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] > > > > > > > > __
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/aster> isk-users > > > To > > > UNSUBSCRIBE or update options visit: > > > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > ___ > > 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 > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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/aster> isk-users > > To > > UNSUBSCRIBE or update options visit: > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > 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 ___ 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
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/aster> isk-users > To > UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users ___ 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
[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
