[asterisk-users] Call problems

[Emmanuel Pascal Bruno]

I have a DID from IPKall.com which is forwarded to my asterisk box.
Then this extension should call my ip phone using Dial application.
Everything works fine, except when I pickup the phone, I can talk, the other
party can hear me, but I cannot hear anything the person says on the ip
phone.
Then after a couple of seconds, the call hangs up.  I don't know why.

Here is the message I get:

 SIP/ipphone-09401f10 answered SIP/XX.XX.XXX.XX-09400918
    -- Native bridging SIP/XX.XX.XXX.XX-09400918 and SIP/ipphone-09401f10
[Oct 30 13:38:12] WARNING[7290]: chan_sip.c:2787 retrans_pkt: Maximum
retries exceeded on transmission
4056be591b329cc9441f75b4560c3ccb at XX.XX.XXX.XX<4056be591b329cc9441f75b4560c3ccb at 66.54.140.46>for
seqno 102 (Critical Response) -- See doc/sip-retransmit.txt.
[Oct 30 13:38:12] WARNING[7290]: chan_sip.c:2814 retrans_pkt: Hanging up
call 4056be591b329cc9441f75b4560c3ccb at XX.XX.XXX.XX<4056be591b329cc9441f75b4560c3ccb at 66.54.140.46>-
no reply to our critical packet (see doc/sip-retransmit.txt).
  == Spawn extension (ipkall, ipphone, 1) exited non-zero on
'SIP/XX.XX.XXX.XX-09400918'

I am running asterisk 1.6 on CentOS

Please help me fix this
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