[asterisk-users] asterisk 1.6.1.0 and dial plan changes

[David Backeberg]

On Fri, May 29, 2009 at 1:22 AM, Tharanga <tharanga at roomsnet.com> 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.