Re: [Asterisk-Users] Call quality problems
Michael Welter wrote: I'm having difficulty with an Asterisk system. The external party has very good call quality, but the internal party hears clipping and drop outs. RX Gains too high IRQ sharing of the of the ZAP device High load of the machine Are a few that come to mind. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call quality problems
Doug Lytle wrote: Michael Welter wrote: I'm having difficulty with an Asterisk system. The external party has very good call quality, but the internal party hears clipping and drop outs. RX Gains too high IRQ sharing of the of the ZAP device There is no ZAP device (it is a SIP-only implementation) and there are no interrupts being shared. High load of the machine The machine is totally idle. The T1 vendor noticed 2% packet loss during a ping flood originating from outside. We changed the Cisco IAD, and there is no longer packet loss, but we still have the clipping. Asterisk 1.2.4. -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 [EMAIL PROTECTED] www.TelecomMatters.net ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call quality problems
Michael Welter wrote: Doug Lytle wrote: Michael Welter wrote: The machine is totally idle. The T1 vendor noticed 2% packet loss during a ping flood originating from outside. We changed the Cisco IAD, and there is no longer packet I've noted from employees that the volumes levels on the phones themselves, when set too high will cause crackling. Does the crackling coincide with talking on the local side? What firmware are you running on the Polycoms? Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call quality problems
Doug Lytle wrote: Michael Welter wrote: Doug Lytle wrote: Michael Welter wrote: The machine is totally idle. The T1 vendor noticed 2% packet loss during a ping flood originating from outside. We changed the Cisco IAD, and there is no longer packet I've noted from employees that the volumes levels on the phones themselves, when set too high will cause crackling. Does the crackling coincide with talking on the local side? What firmware are you running on the Polycoms? I'm not on site, but I remember 1.6.4. It's not really crackling or popping that's the problem. The problem is with dropouts. It also seems that the trailing edge of each word will sometimes be lost (possibly a dropout). If you're familiar with the WWV time signal (303-499-7111), for the first 45 minutes of each hour there is a tone interrupted by a click every second (during the last 15 minutes it's just the clicks). When I listen to this on the Asterisk system, the tone only lasts for a fraction of a second and then silence until the next click. Is the phone (or Asterisk) performing echo suppression that drops the last part of the tone? Also, there are no ZAP cards in the system. What timing source does SIP use to play the incoming media stream? Thanks for your comments, Doug. -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 [EMAIL PROTECTED] www.TelecomMatters.net ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call quality problems
Michael Welter wrote: I'm not on site, but I remember 1.6.4. I had in place 1.6.2, and had way to many problems with it. I reverted back to 1.5.2 and things cleared up. Is the phone (or Asterisk) performing echo suppression that drops the last part of the tone? I believe the phone does some E.C. along with Asterisk. Also, there are no ZAP cards in the system. What timing source does SIP use to play the incoming media stream? I think the only time you need a timing source is if you are mixing audio streams, i.e. meetme, MOH. In which case you'd probably need to run ztdummy. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call quality problems
Doug Lytle wrote: I think the only time you need a timing source is if you are mixing audio streams, i.e. meetme, MOH. In which case you'd probably need to run ztdummy. Yes , ztdummy is running. I'm going to (temporarily) put a TDM card in the system just to eliminate that possibility. -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 [EMAIL PROTECTED] www.TelecomMatters.net ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call quality problems
I'm having difficulty with an Asterisk system. The external party has very good call quality, but the internal party hears clipping and drop outs. The WAN comes in from the Cisco IAD and into a LAN switch (DLink DGS-1005D w/ 802.1p) where the two public IPs are switched to different devices. One device is a FireBox device controlling a separate LAN with VPNs. The other device is eth0 on the Asterisk system. On the Asterisk eth1 is a 3Com 2226 LAN switch which connects Polycom IP501 phones. There are no PCs on this voice LAN. All ports on all LAN switches indicate full duplex. The quality problem doesn't appear to be volume related (a single call still has problems). The Polycom IP501s use SIP to the PBX, and the PBX uses SIP to the provider. The normal WWV time signal consists of a constant tone that is interrupted every second by a click. On the Polycom, each click can be heard, the tone starts, but the tone is clipped and there is silence until the next click. I've verified that QoS is enabled in the IAD. I would appreciate your thoughts. Thanks, -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 [EMAIL PROTECTED] www.TelecomMatters.net ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
