Asterisk 1.4 and xlite problem

Hi all,

I have a problem using asterisk 1.4 with xlite3.0

all equipment are in the same subnet(asterisk&xlite) no nat involve in this case and I’d like to bypass RTP stream from asterisk server so I used

canreinvite=yes

In sip.conf file , I add users like

[1111]
type=friend
callerid=“user01” <1111>
host=dynamic
canreinvite=yes
username=1111
secret=user01
disallow=all
allow=gsm
allow=ulaw
allow=alaw
dtmfmode=info

[1112]
type=friend
callerid=“user02” <1112>
host=dynamic
canreinvite=yes
username=1112
secret=user02
disallow=all
allow=gsm
allow=ulaw
allow=alaw
dtmfmode=info

1111 and 1112 can call each other but when call setup established both of them cannot hear each other

but if I used canreinvite=no in 1111,1112 everything is ok
but as I say I’d to bypass RTP stream and It should be work in this case because all device are in the same subnet

Anyone has an idea please help me
Thanks in advance