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@watertech.com
Do you have ztdummy loaded?> -----Original Message----- > From: Chris Shaw [mailto:chriss@watertech.com] > Sent: Friday, July 09, 2004 1:14 PM > To: asterisk-users@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@watertech.com > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > lists.digium.com/mailman/listinfo/aster> isk-users > To > UNSUBSCRIBE or update options visit: >lists.digium.com/mailman/listinfo/asterisk-users