[asterisk-users] asterisk 1.6.1.0 and dial plan changes

[Tharanga]

Hi all,

I have installed asterisk latest stable version 1.6.1.0, with dahdi 
driver (tdm410p). then i try to use my older 1.4 extensions.conf. . now 
it wont work with 1.6.

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.

please help me to solve the issue. all work on asterisk 1.4.

[general]

port = 5060                    
bindaddr = 0.0.0.0          
context = sip       
disallow=all
allow=all
;allow=g729
;allow=gsm
allow=alaw
allow=ulaw
transfer=yes
tos=lowdelay
dtmfmode = rfc2833

[312]
type=friend                    ; Friends place calls and receive calls
context=sip2            ; Context for incoming calls from this user
secret=312
host=dynamic                ; This peer register with us
dtmfmode=rfc2833               ; Choices are inband, rfc2833, or info
username=312               ; Username to use in INVITE until peer registers
mailbox=312
qualify=yes
disallow=all
pickupgroup=1
allow=all
;allow=alaw                     ; dtmfmode=inband only works with ulaw 
or alaw!
;allow=gsm
;;canreinvite=no
;;progressinband=yes
;;reinvite=no
;;callerid=tharanga <312>


extensions.conf



channel.dadhi.conf

[channels]


signalling=fxs_ks
;toneduration=100
callwaiting=yes
threewaycalling=yes
callreturn=yes
echocancel=128,param1=32,param2=0,param3=14
echocancelwhenbridged=yes
echotraining=yes
echotraining=800
busydetect=yes
busycount=2
hanguponpolarityswitch=yes
ringtimeout=8000
group=1
context=sip
immediate=yes
jitterbuffers=4
jbenable = yes
echocancel=yes
channel=>1-4
;overlapdial=yes
;pulsedial=yes
dtmfmode=rfc2833
;relaxdtmf=yes
;rxgain=10.0
;txgain=8.0


Many thanks
Tharanga