On Fri, May 29, 2009 at 1:22 AM, Tharanga <[email protected]> wrote: > I managed to register my phone on asterisk. but i cant hear any dial > tone on my phone. these are my configs. it will detect incoming calls > and transfer the call to ext 312. but sip phone users voice is not > clear..., but sip phone user can hear the other party (PSTN) very clearly.
You've mentioned like three different things, each of which you should attack separately. I can give some tips on the SIP voice quality issue: * take a look at dsp.conf, and make a larger silencethreshold value. I set mine to 1000. * take a look at codecs.conf, and change vad => false You don't say the kind of call you're making, but if you're using MeetMe() I have more advice regarding voice quality with conference rooms. _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
