I'm trying to use Asterisk with a TDM400P and 2 analog lines, but I'm having a hard time getting the kind of audio quality that I'd like.
I'm hoping to be able to use SIP phones to make calls through Asterisk and have the same quality as a regular analog phone connected to the PSTN. Are my expectations too high? Is it that once I'm using Asterisk as a gateway to the PSTN, I have to expect a certain amount of voice quality degradation? I've never used Asterisk before so I really don't know what is truly possible and what isn't. I'm getting a slight echo...sometimes...it varies from call-to-call, but the biggest problem I have is a constant hiss in the background. Again, this varies from call-to-call. I know my SIP phones are fine as SIP-to-SIP calls on my LAN work perfectly. I only have problems going out to the PSTN. So for those of you isng a TDM400P, would you say your voice quality is as good as a pure analog/PSTN call? I have done the normal troubleshooting that I've seen recommended like changing SIP phones, ensuring that the IRQ isn't shared, tweaking the tx and rx gain and the echo cancellation values. I haven't tried EVERY possible combination of values, but if what I want isn't achievable, then I don't want to waste my time. _______________________________________________ --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
