Bridging channels between zapata and sip

Hi to one and all

i, working a trainee.
My task is to bridge channel between sip and zapata

Already Active call exist between TWO sip phones (say A &B ,Here A called B), now i created one more new channel by dialing from PSTN through asterisk , here i want to bridge between this PSTN phone & to one of SIPphone B.
My task is to bridge 2 channels (zapata and sip) without droping call(B)
i am able to bridge these two channels , i am able to hear voice in sip phone when one talks from pstn but pstn user on other side is not able to hear my voice (when talking from sip phone)

in logs displays a message like

unable to grab the voice frames from the sip user[xxxx]

i am working on this problem from 1 week , can any one suggest me hw can i approach to solve this problem !!
let me know different approaches to bridge two channels of different technologies

Thanks in Advance to one and all

Have you run
sip debug
on the asterisk CLI for information on SIP packets transmitted.

Audio from one side also can be because of NAT/firewall problems.
In that case you can search this forum with the releveant terms.