Problem after upgrading asterisk to 10.6-7 from 10.5

We have been using asterisk 10.5.0 for quite some time on our dedicated server.

Recently we moved to asterisk 10.6 and then to 10.7 respectively.

We are no longer to route audio in the way it should have been routed using directmedia.

Following is our sip.conf :

[general]
context = default ; Default context for incoming calls
allowoverlap=no
udpbindaddr=0.0.0.0
tcpenable=no
tcpbindaddr=0.0.0.0
srvlookup=yes
videosupport=no
dtmfmode=rfc2833
nat=yes
directmedia=yes
directrtpsetup=yes
disallow=all
allow=g729
alwaysauthreject=yes
rtcachefriends=yes
sendrpid=yes
trustrpid=yes
insecure=port,invite
tos=0x14
tos_sip=cs3
tos_audio=ef
cos_sip=3
cos_audio=5
limitonpeers=yes
limitonpeer=yes
callevents=yes
notifyringing=yes
notifyhold=yes
preferred_codec_only=yes
use_q850_reason=yes
progressinband=never
prematuremedia=yes
externip=176.31.228.131
qualify=yes

[voiprovider1]
type=peer
defaultuser=username
secret=password
host=sip.voipprovider.com

We are using a2billing as our VOIP gateway. Directmedia seems to work absolutely fine till 10.5 but not beyond. Are we doing something wrong over here?

Please help if anybody is facing the same issue. The audio is getting disconnected after 1-2 seconds.

Regards,
Debanjan Basu.
(Founder toRing VOIP)

45 views and still no solution.

Please help me ASAP.
We are really looking forward to upgrading our asterisk server to 10.7 from 10.5.

There is no debugging information.

Ours is a production server we terminate more than 3000+ channels simultaneously.

Only on Sunday can I post the SIP Debug for the calls.

Please help me Sir, in due course.