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In iax.conf, do you have jitterbuffer set, try jitterbuffer=no, or try some values from 1-5, i use 3 -----Original Message----- From: Jan Rychter <jan@rychter.com> To: asterisk-users@lists.digium.com <asterisk-users@lists.digium.com> Date: July 16, 2003 11:45 AM Subject: [Asterisk-Users] IAX pauses >Hi, > >I'm running asterisk in the following setup > >Phone -> WX100USB -> * -> Internet -> * ->...

I'm having trouble with a WX100USB adapter and a Siemens Gigaset cordless phone. If I select fxols as a signalling method, calls are being disconnected. Usually after about 4 minutes, and asterisk just says that the phone has hung up. If I choose fxogs, I immediately get a LINE IN USE message on my phone and I can't even get a dialtone. If I choose fxoks, it mostly works, but sometimes

Does G.729 provide better voice quality than GSM? (a question for people who have tried both) --J. -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 188 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20030715/52f18205/attachment.pgp

Hi, I'm looking for a good codec to use on a personal VoIP setup. It is strictly for my personal use, I'll never resell it, make money or it, or whatever. It seems a free personal-use G729 codec is available as a WIN32 library. I find it puzzling that at the same time one has to pay license fees to use it under Linux, even non-commercially. I was wondering -- am I missing something?

This could be a bug in the e1071 svm code, but maybe not -- I guess I'll send it here anyway. It's reproducible. > x <- seq (0.1,5,by=0.05) > y <- log(x) + rnorm (x, sd=0.2) > library(e1071) > m <- svm (x,y) Process R segmentation fault at Tue Nov 20 23:34:19 2001 > version _ platform i686-pc-linux-gnu arch i686 os

Seeing that many people here hit problems with activating their G.729 licenses, I decided to post my opinion. I have purchased two G.729 licenses, for my private use. I did this even though VoiceAge makes G.729 free for private use, as Windows libraries. I guess a sufficiently motivated person could take the COFF libraries, run them through objcopy on cygwin (producing ELF .o files) and link them

I have hit a problem where chan_h323 sometimes doesn't hang up properly and stays stuck in the "Up" state, with asterisk consuming 100% of CPU: *CLI> show channels Channel (Context Extension Pri ) State Appl. Data H323/ip$127.0.0.1:30008/21552 (local 123 1 ) Up (None) (None) 1 active channel(s) *CLI>

...7 Aug 2001 22:42:07 +0200 (MET DST) * PR# 1116 * Subject: get.hist.quote does not work From: arto.luoma@uta.fi Date: Thu, 4 Oct 2001 14:45:29 +0200 (MET DST) --Seems specific to one locale on one system. --The bug is unlikely to be in R. * PR# 1178 * Subject: segfault using svm from e1071 From: Jan Rychter <jan@rychter.com> Date: Tue, 20 Nov 2001 23:38:17 +0100 * PR# 1199 * Subject: pixmap: infinite recursion with nonascii pnm-files From: thomas.baumann@ch.tum.de Date: Fri, 7 Dec 2001 11:07:52 +0100 (CET) * PR# 1295 * Subject: typo and user-proofing in odesolve() From: Ben Bolker <bolker@z...

...g in getCovariateFormula From: Setzer.Woodrow@epamail.epa.gov Date: Tue, 31 Jul 2001 11:24:09 -0400 --change needed in package nlme * PR# 1044 * Subject: Polymarsall.c From: pleu@hotmail.com Date: Tue, 7 Aug 2001 22:42:07 +0200 (MET DST) * PR# 1178 * Subject: segfault using svm from e1071 From: Jan Rychter <jan@rychter.com> Date: Tue, 20 Nov 2001 23:38:17 +0100 * PR# 1199 * Subject: pixmap: infinite recursion with nonascii pnm-files From: thomas.baumann@ch.tum.de Date: Fri, 7 Dec 2001 11:07:52 +0100 (CET) * PR# 1295 * Subject: typo and user-proofing in odesolve() From: Ben Bolker <bolker@z...

...g in getCovariateFormula From: Setzer.Woodrow@epamail.epa.gov Date: Tue, 31 Jul 2001 11:24:09 -0400 --change needed in package nlme * PR# 1044 * Subject: Polymarsall.c From: pleu@hotmail.com Date: Tue, 7 Aug 2001 22:42:07 +0200 (MET DST) * PR# 1178 * Subject: segfault using svm from e1071 From: Jan Rychter <jan@rychter.com> Date: Tue, 20 Nov 2001 23:38:17 +0100 * PR# 1199 * Subject: pixmap: infinite recursion with nonascii pnm-files From: thomas.baumann@ch.tum.de Date: Fri, 7 Dec 2001 11:07:52 +0100 (CET) * PR# 1295 * Subject: typo and user-proofing in odesolve() From: Ben Bolker <bolker@z...

...g in getCovariateFormula From: Setzer.Woodrow@epamail.epa.gov Date: Tue, 31 Jul 2001 11:24:09 -0400 --change needed in package nlme * PR# 1044 * Subject: Polymarsall.c From: pleu@hotmail.com Date: Tue, 7 Aug 2001 22:42:07 +0200 (MET DST) * PR# 1178 * Subject: segfault using svm from e1071 From: Jan Rychter <jan@rychter.com> Date: Tue, 20 Nov 2001 23:38:17 +0100 * PR# 1199 * Subject: pixmap: infinite recursion with nonascii pnm-files From: thomas.baumann@ch.tum.de Date: Fri, 7 Dec 2001 11:07:52 +0100 (CET) * PR# 1295 * Subject: typo and user-proofing in odesolve() From: Ben Bolker <bolker@z...

I've tried making calls using the console (both ALSA and OSS). ALSA seems to work after applying the little fix posted on this list some time ago by someone (which I'll submit into the bug tracker), but all I get is one-way audio: I can hear the other end, but nothing gets transmitted. At first I thought this was an audio problem, but it doesn't seem to be. My machine isn't

[I have sent a message about SIP problems via gmane, but it seems the list is gatewayed one-way only...] The message was: I've been trying to use Asterisk as a SIP->PSTN gateway. It runs fine when the SIP client is on the local network and there is not packet loss. But now I've tried running a remote client (halfway around the globe) -- this works great until some packets get lost.

What is the state of speex support in asterisk? I saw the codec seems to be there. Can speex be used on IAX2 links? Is there much work still to be done? many thanks, --J.