On Thu, 20 May 2004 14:18:16 +0100, "Andy Farnsworth" <[EMAIL PROTECTED]> wrote:
>As a test, I was trying to use Iaxcomm and Iaxphone to connect to >Asterisk and dial out to my other line. Using either of these soft >phones, I can connect to Asterisk and listed to audio just fine. I can >even connect across the net to another asterisk server and hear audio >just fine, however, when I dial out to my second land line the audio >that is transmitted is horribly broken up. It is as if the audio stream >is broken into 8 parts every second and then every other part is >dropped. I then tried the asterisk echo test and got the same thing. I >am running Asterisk under RH9 on an AMD 2600+ 512 Mb RAM Desktop and the >soft phones on my laptop running Windows XP (Laptop is Sony Vaio >PCG-GRT815E, 2.8 Ghz processor, 512Mb Ram). > >Is this an asterisk problem or a soft phone problem? If asterisk, any >ideas on how to fix it? What kind of PSTN interfaces are you using? I'm not sure from the description: are you seeing the problem of both lines, or only the second line? Do you get the same kind of results when using a SIP softphone? I'll be posting new binaries to sourceforge this weekend, because there have been some library changes related to jitter, but I haven't heard or seen anything as drastic as you describe. BTW what version of asterisk? > >Thanks, > >Andy Farnsworth > > >_______________________________________________ >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 _______________________________________________ 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