Kris Boutilier
2004-Sep-07 17:55 UTC
[Asterisk-Users] Maximum tollerable lag/jitter for IAX2 w/o j itterbuffer enabled?
Unfortunatly no on both counts. The arrangement right now has: PSTN Trunks & Stations <-> Nortel Norstar#1 <-CT1-> Asterisk#1 <-IAX2-> Asterisk#2 <-CT1-> Nortel Nortstar#2 <-> Stations The Asterisk boxes provide Voicemail to their sites Norstars and intersite calls over IAX. Local Voicemail works flawlessly at each site but there have been reports of PSTN calls that are in voicemail on Asterisk#2 being cut off mid-call. Likewise callers from PSTN to stations on Norstar#2 have occasionally reported dropped calls. There have been no reports of problems with intersite calls from Norstar#1 to Norstar#2, Norstar#1 to Asterisk#2 Voicemail or PSTN to Asterisk #1 Voicemail. Which led me to an IAX problem... The only other possibly related symptom might be far-end echo experienced on the station side of Norstar#2 to long distance numbers on the PSTN. I have: echocancel=yes and echocancelwhenbridged=yes enabled in zapata.conf on both hosts but am still using the default echo canceller code. This is 'Asterisk CVS-HEAD-08/13/04-10:37:13'> -----Original Message----- > From: Eric Wieling [mailto:eric@fnords.org] > Sent: September 7, 2004 5:43 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [Asterisk-Users] Maximum tollerable lag/jitter for IAX2 > w/oji tterbuffer enabled? > > > You're not using callprogress=yes or busydetect=yes are you? > That could > easily cause these problems with Zap ports. >{clip}
Kris Boutilier
2004-Sep-07 23:07 UTC
[Asterisk-Users] Maximum tollerable lag/jitter for IAX2 w/o j itterbuffer enabled?
> -----Original Message----- > From: Benjamin on Asterisk Mailing Lists > [mailto:benjk.on.asterisk.ml@gmail.com] > Sent: September 7, 2004 10:53 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Maximum tollerable lag/jitter > for IAX2 w/o jitterbuffer enabled? > > > On Tue, 7 Sep 2004 16:26:24 -0700, Kris Boutilier > <kris.boutilier@scrd.bc.ca> wrote: > > I'm having a problem with intersite calls over IAX2 being abruptly > > terminated. Nothing odd shows in any of the logs for > Asterisk or the host. > > The only think I can think it might be is a lag-spike on > the site to site > > connection. > > When does the cut off occurr? Is it always after about 8-10 seconds? > If so, you may have a problem with IAX transfer. You can verify this > by using notransfer=yes. >Having notransfer=no is beginning to sound like it's the culprit, but I don't understand why. I don't see any related issues in bugs.digium.com nor is there any intervening network equipment that would interfere with the transit of packets between hosts (such as a NAT firewall). The only thing that springs to mind, if it's all UDP driven, is a lack of retry handler for the UDP handoff acknowledgement? I'm averaging about a 0.5% collision rate on this network (half-duplex 10Base-T)... Kris Boutilier Information Systems Coordinator Sunshine Coast Regional District
Andrew Kohlsmith
2004-Sep-08 06:24 UTC
[Asterisk-Users] Maximum tollerable lag/jitter for IAX2 w/o j itterbuffer enabled?
On Tuesday 07 September 2004 20:55, Kris Boutilier wrote:> The arrangement right now has: > PSTN Trunks & Stations <-> Nortel Norstar#1 <-CT1-> Asterisk#1 <-IAX2-> > Asterisk#2 <-CT1-> Nortel Nortstar#2 <-> Stations> The Asterisk boxes provide Voicemail to their sites Norstars and intersite > calls over IAX. Local Voicemail works flawlessly at each site but thereHow are you getting MWI to light up on the digital phones? I was going to start screwing about with MCDN to try and achieve this (turning the MICS into nothing more than a digitla phone driver) but haven't started yet. -A.
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