Hi Dan, Steve, Michael, Bruno and others. I will try to describe my VoIP environment below:
SERVER: - FC1 with Asterisk CVS-v1-0-11/04/04-23:47:17 - iax.conf [general] bindport = 4569 bindaddr = 0.0.0.0 delayreject=yes disallow=all allow=ulaw allow=alaw allow=gsm tos=lowdelay jitterbuffer=no dropcount=2 maxjitterbuffer=100 maxexccessbuffer=100 mailboxdetail=yes [1001] callerid="Ramal 1001" <1001> context=from-internal host=dynamic mailbox=1001 notransfer=yes port=4569 secret=**** type=friend username=1001 [1002] callerid="Ramal 1002" <1002> context=from-internal host=dynamic mailbox=1002 notransfer=yes port=4569 secret=**** type=friend username=1002 CLIENT 1001: - Windows XP - DIAX 0.9.9g - Firefly 1.9.6 Build 3944 - USB Phone NTP200E - Compatible with ATCOM USB Phone - AMD 1.8Ghz with 256Mb CLIENT 1002: - Windows XP - DIAX 0.9.9g - Firefly 1.9.6 Build 3944 - USB Phone NTP200E - Compatible with ATCOM USB Phone - AMD 1.66Ghz with 256Mb ADDITIONAL INFORMATION - All machines are in the same network(192.168.*.*) no firewall in the middle; - With Firefly I have a VERY GOOD conversation, without any delay; - With DIAX I have a one way delay of 10 sec. Only the person who recieve the call get the delay, the person who make the call listen without problems; - Firefly in one side and DIAX in the other side, same delay problem; - No problems with SIP; - No problems(delay) with Linux clients runnig IaxComm 0.99pre11; - Same problem with DIAX oldest DLL; - Ping from clients to server: 0% packet loss and < 1ms; - No problems calling PSTN, Voicemail, etc, just between DIAX clients; If you need something else, let me know! Thanks for your help! Denis Galv�o. Em Dom 16 Jan 2005 19:52, Steve Kann escreveu: > On Jan 16, 2005, at 2:53 PM, Dan wrote: > > Hi Steve, > > > > ----- Original Message ----- From: "Steve Kann" <[EMAIL PROTECTED]> > > > >> On Jan 14, 2005, at 2:03 PM, Dan wrote: > >>> Hi, > >>> > >>> \> Em Sex 14 Jan 2005 16:43, Dan escreveu: > >>>>> > I dont have problems when calling PSTN extensions, and calling > >>>>> > VoceMail, EchoTest, etc. The problem is related with the > >>>>> > >>>>> conversation > >>>>> > >>>>> > between two DIAX Softphones. > >>>>> > >>>>> Between 2 DIAX phone and the delay is in one direction only?? > >>>> > >>>> Yes. One direction only... Just who make the call get the delay. > >>> > >>> Then try > >>> jitterbuffer=no > >>> in iax.conf > >>> to see if it solves this issue. > >> > >> Dan et. al, > >> I think this might be a problem with native transfers, and needing to > >> reset the jitterbuffer history when this happens, or something like > >> this.. > >> -SteveK > > > > But I have tried and I do don't have this problem here... > > What can I do to make this happen here? > > I don't know... > > Maybe if we could get a packet trace of the situation that causes the > problem? > > Maybe try notransfer or whatever the iax.conf parameter is, and see if > that changes things. If it does, it points towards this being the > problem. > > If the delay goes down after a couple of minutes after the transfer, > this could be the problem. If it doesn't, there's something else > really wrong.. > > (I'm assuming you're using the new JB code here..). Also, if you're > using the new JB code, you should implement the stuff to get the > network stats, so we can see if calculated jitter is substantially > higher..) > > _______________________________________________ > Asterisk-Users mailing list > [email protected] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- D e n i s G a l v � o iSolve - Solve Is Our Business Av. Candido de Abreu, 526 1206B CEP: 80530-000 - Curitiba - PR +55 41 252-2977 http://www.isolve.com.br _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
