RE: [asterisk-users] Sangoma A102 w/ EC module gets intermittent echo/audio artifacts --followup and resolution
Followup on this issue, it appears that using a single PRI's clock as the master clock avoids clock drift between the PRI's and we get no more artifacts. So, : wanpipe1.conf: TE_CLOCK= NORMAL TE_REF_CLOCK= 0 wanpipe2.conf: TE_CLOCK= MASTER TE_REF_CLOCK= 1 zaptel.conf: span=1,1,0,esf,b8zs span=2,0,0,esf,b8zs -Original Message- From: Michael L. Young [mailto:[EMAIL PROTECTED] Sent: Wednesday, January 03, 2007 9:37 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Sangoma A102 w/ EC module gets intermittent echo/audio artifacts Zaptel.conf: loadzone = us span=1,1,0,esf,b8zs span=2,1,0,esf,b8zs bchan=1-23 dchan=24 bchan=25-47 dchan=48 Just a quick thought in looking at the settings above, it appears that you have set both spans as the primary timing source. I am pretty sure that only one span should be the primary timing source. The other span should either be at 0 (not used as a timing source) or set as a secondary timing source. Hope this helps. Michael L. Young ___ --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 ___ --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 ___ --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] Sangoma A102 w/ EC module gets intermittent echo /audio artifacts
I've replaced 2XTE110 with an A102 with echo cancellation specifically to deal with echo problems. However, user feedback has indicated to me that on some calls (not a lot, but some) the call is unusable, with audio artifiacts, described by one user, as: very bad phasing reverb feedback (from my rock roll days). This is quite intermittent, as in most cases, the user says, it's a one-time thing, they hang up, the problem caller calls back, everythings good. It's as if the Sangoma is trying too hard? I personally have not heard this, but I have to trust what the users say. Some ideas I'd like to bounce: 1. tx and rxgains - this card is plugged into an Atlas 550 which seems to run a little hot on the gains. 2. Timing - the card takes it's sync from the Atlas, which in turn syncs from the PRI. Maybe have one port on the card take it's timing from the other port? Don't see how it would be relevant, but hey, that's all I've got. 3. Taps? Does an A102 even care about taps or just echocancel=yes? Running * 1.0.9, Zaptel 1.0.9, FC2 yum-updated to current, quad Xeon. Production box, handles 2-6 thousand calls a day, snom 360 handsets w/ latest firmware, load average never goes over 2.0 My conf files: wanpipe1.conf: [devices] wanpipe1 = WAN_AFT_TE1, Comment [interfaces] w1g1 = wanpipe1, , TDM_VOICE, Comment [wanpipe1] CARD_TYPE = AFT S514CPU = A CommPort= PRI AUTO_PCISLOT= NO PCISLOT = 4 PCIBUS = 1 FE_MEDIA= T1 FE_LCODE= B8ZS FE_FRAME= ESF FE_LINE = 1 TE_CLOCK= NORMAL TE_REF_CLOCK= 0 TE_HIGHIMPEDANCE= NO LBO = 0DB FE_TXTRISTATE = NO MTU = 1500 UDPPORT = 9000 TTL = 255 IGNORE_FRONT_END = NO TDMV_SPAN = 1 TDMV_DCHAN = 0 [w1g1] ACTIVE_CH = ALL TDMV_ECHO_OFF = NO TDMV_HWEC = YES wanpipe2.conf: [devices] wanpipe2 = WAN_AFT_TE1, Comment [interfaces] w2g1 = wanpipe2, , TDM_VOICE, Comment [wanpipe2] CARD_TYPE = AFT S514CPU = A CommPort= PRI AUTO_PCISLOT= NO PCISLOT = 4 PCIBUS = 1 FE_MEDIA= T1 FE_LCODE= B8ZS FE_FRAME= ESF FE_LINE = 2 TE_CLOCK= NORMAL TE_REF_CLOCK= 0 TE_HIGHIMPEDANCE= NO LBO = 0DB FE_TXTRISTATE = NO MTU = 1500 UDPPORT = 9000 TTL = 255 IGNORE_FRONT_END = NO TDMV_SPAN = 2 TDMV_DCHAN = 0 [w2g1] ACTIVE_CH = ALL TDMV_ECHO_OFF = NO TDMV_HWEC = YES Zaptel.conf: loadzone = us span=1,1,0,esf,b8zs span=2,1,0,esf,b8zs bchan=1-23 dchan=24 bchan=25-47 dchan=48 Zapata.conf: [channels] language=en context=from-pstn switchtype=national pridialplan=unknown signalling=pri_cpe faxdetect=incoming usecallerid=yes echocancel=yes echocancelwhenbridged=yes txgain=-2.0 rxgain=-2.0 group=0 channel = 1-23 channel = 25-47 ___ --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] Sangoma A102 w/ EC module gets intermittent echo /audio artifacts ---More information
Aha, it just happened to me, so now I can characterize the audio: It basically sounds like it's missing every other sample - fuzzy and distorted. Timing? ___ --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] Sangoma A102 w/ EC module gets intermittent echo/audio artifacts
Zaptel.conf: loadzone = us span=1,1,0,esf,b8zs span=2,1,0,esf,b8zs bchan=1-23 dchan=24 bchan=25-47 dchan=48 Just a quick thought in looking at the settings above, it appears that you have set both spans as the primary timing source. I am pretty sure that only one span should be the primary timing source. The other span should either be at 0 (not used as a timing source) or set as a secondary timing source. Hope this helps. Michael L. Young ___ --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] Sangoma A102 w/ EC module gets intermittent echo/audio artifacts
I think you are absolutely right. The audio I heard earlier sounds exactly like a timing issue. So: wanpipe1.conf: TE_CLOCK= NORMAL TE_REF_CLOCK= 0 wanpipe2.conf: TE_CLOCK= MASTER TE_REF_CLOCK= 1 zaptel.conf: span=1,1,0,esf,b8zs span=2,0,0,esf,b8zs I'm going to make this change and reload at lunchtime, I'll document it and post it to the list if it works. thanks for the good eye. -Original Message- From: Michael L. Young [mailto:[EMAIL PROTECTED] Sent: Wednesday, January 03, 2007 9:37 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Sangoma A102 w/ EC module gets intermittent echo/audio artifacts Zaptel.conf: loadzone = us span=1,1,0,esf,b8zs span=2,1,0,esf,b8zs bchan=1-23 dchan=24 bchan=25-47 dchan=48 Just a quick thought in looking at the settings above, it appears that you have set both spans as the primary timing source. I am pretty sure that only one span should be the primary timing source. The other span should either be at 0 (not used as a timing source) or set as a secondary timing source. Hope this helps. Michael L. Young ___ --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 ___ --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] Sangoma A102 w/ EC module gets intermittent echo /audio artifacts
Colin Anderson wrote: I've replaced 2XTE110 with an A102 with echo cancellation specifically to deal with echo problems. However, user feedback has indicated to me that on some calls (not a lot, but some) the call is unusable, with audio artifiacts, described by one user, as: very bad phasing reverb feedback (from my rock roll days). This is quite intermittent, as in most cases, the user says, it's a one-time thing, they hang up, the problem caller calls back, everythings good. It's as if the Sangoma is trying too hard? I personally have not heard this, but I have to trust what the users say. Some ideas I'd like to bounce: 1. tx and rxgains - this card is plugged into an Atlas 550 which seems to run a little hot on the gains. 2. Timing - the card takes it's sync from the Atlas, which in turn syncs from the PRI. Maybe have one port on the card take it's timing from the other port? Don't see how it would be relevant, but hey, that's all I've got. 3. Taps? Does an A102 even care about taps or just echocancel=yes? I have fixed similar problems by reducing the gains. I call it ECFO Echo Cancel Freak Out. ___ --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
