[OpenSIPS-Users] no ringback

[Jinsong Hu]

Hi, There:
  I am using opensips/kamailio in front of asterisk pool. my user register 
on the opensips, and pstn call are routed out via asterisk.  what I find out 
is that when the caller calls callee, some of the UA doesn't generate ring 
back. for example, if I use xlite, the ring back works fine. but if I use 
sipura 3000,
I don't hear anything until the callee picks up phone.
  I did a debug and found that after INVITE, I get 200 back, and then the UA 
sends out ACK. the callee never sends 180 or 183 back to the caller UA. so 
before the callee pick up phone, all the caller can hear is just silence.

  if my user registers directly on the asterisk, he can hear the ringback 
because the Dial() command by default
will send ring back to the UA.

  How do I solve this problem in this case ? I searched all over internet 
and don't see any body having any solution.

Jimmy